vicidial.org

Posted: **Wed May 06, 2020 9:25 am**

Hi
I am into a strange issue, completed a new vicibox installation. When I try to dial manual from agent screen call is connecting fine and answered, but after dial time out it shows Agent Alert on agent interface that "Dial timed out, contact your system administrator" but the call is still connected. below is the CLI output for the manual call I mentioned. and also there is no hangup output at the CLI after callee hangs up the call.

Code: Select all: [May 6 09:59:18] -- Called 8600051@default [May 6 09:59:18] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-000000c4;2", "8600051,F") in new stack [May 6 09:59:18] -- Local/8600051@default-000000c4;1 answered [May 6 09:59:18] -- Executing [91601XXXXXXX@default:1] AGI("Local/8600051@default-000000c4;1", "agi://127.0.0.1:4577/call_log") in new stack [May 6 09:59:18] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=001)) [May 6 09:59:18] -- <Local/8600051@default-000000c4;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [May 6 09:59:18] -- Executing [91601XXXXXXX@default:2] Set("Local/8600051@default-000000c4;1", "CALLERID(ALL)="" <8336043243>") in new stack [May 6 09:59:18] -- Executing [91601XXXXXXX@default:3] Dial("Local/8600051@default-000000c4;1", "SIP/1601XXXXXXX@zsip,60,Tto") in new stack [May 6 09:59:18] == Using SIP RTP CoS mark 5 [May 6 09:59:18] -- Called SIP/1601XXXXXXX6@zsip [May 6 09:59:19] == Manager 'sendcron' logged off from 127.0.0.1 [May 6 09:59:20] -- SIP/zsip-000000c1 is ringing [May 6 09:59:21] > 0x7f433800a990 -- Strict RTP learning after remote address set to: 68.68.124.80:26368 [May 6 09:59:21] -- SIP/zsip-000000c1 is making progress passing it to Local/8600051@default-000000c4;1 [May 6 09:59:30] -- SIP/zsip-000000c1 answered Local/8600051@default-000000c4;1 [May 6 09:59:30] -- Channel SIP/zsip-000000c1 joined 'simple_bridge' basic-bridge <c2dd6974-a4bd-4668-9c60-b2c0ef9285c5> [May 6 09:59:30] -- Channel Local/8600051@default-000000c4;1 joined 'simple_bridge' basic-bridge <c2dd6974-a4bd-4668-9c60-b2c0ef9285c5>

below is the output when I press the hangup customer form agent interface.

Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600051@default-000000c4;1
[May 6 10:06:52] -- Channel Local/8600051@default-000000c4;1 left 'simple_bridge' basic-bridge <c2dd6974-a4bd-4668-9c60-b2c0ef9285c5>
[May 6 10:06:52] -- Channel SIP/zsip-000000c1 left 'simple_bridge' basic-bridge <c2dd6974-a4bd-4668-9c60-b2c0ef9285c5>
[May 6 10:06:52] == Spawn extension (default, 91601XXXXXXX, 3) exited non-zero on 'Local/8600051@default-000000c4;1'
[May 6 10:06:52] -- Executing [h@default:1] AGI("Local/8600051@default-000000c4;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----453-----453-----SIP 200 OK)") in new stack
[May 6 10:06:52] -- <Local/8600051@default-000000c4;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... 53-----SIP 200 OK) completed, returning 0
[May 6 10:06:52] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-000000c4;2'
[May 6 10:06:52] WARNING[17925][C-00000156]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[May 6 10:06:52] -- Executing [h@default:1] AGI("Local/8600051@default-000000c4;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[May 6 10:06:52] -- <Local/8600051@default-000000c4;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[May 6 10:06:52] == Manager 'sendcron' logged on from 127.0.0.1
[May 6 10:06:52] NOTICE[18685]: manager.c:4458 action_hangup: Request to hangup non-existent channel: Local/8600051@default-000000c4;2

When i go to autodial mode, outbound calls connecting fine but it disconnects after 8 to 10 sec and no call is passing to the agent logged in. below is the CLI output from autodial mode.

Code: Select all: [May 6 10:11:05] -- Called 91601XXXXXXX@default [May 6 10:11:05] -- Executing [91601XXXXXXX@default:1] AGI("Local/91601XXXXXXX@default-000000c5;2", "agi://127.0.0.1:4577/call_log") in new stack [May 6 10:11:05] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=001)) [May 6 10:11:05] -- <Local/91601XXXXXXX@default-000000c5;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [May 6 10:11:05] -- Executing [91601XXXXXXX@default:2] Set("Local/91601XXXXXXX@default-000000c5;2", "CALLERID(ALL)="" <833XXXXXXX>") in new stack [May 6 10:11:05] -- Executing [91601XXXXXXX@default:3] Dial("Local/91601XXXXXXX@default-000000c5;2", "SIP/1601XXXXXXX@zsip,60,Tto") in new stack [May 6 10:11:05] == Using SIP RTP CoS mark 5 [May 6 10:11:05] -- Called SIP/1601XXXXXXX@zsip [May 6 10:11:06] == Manager 'sendcron' logged on from 127.0.0.1 [May 6 10:11:06] == Manager 'sendcron' logged off from 127.0.0.1 [May 6 10:11:10] -- SIP/zsip-000000c2 is ringing [May 6 10:11:10] -- Local/91601XXXXXXX@default-000000c5;1 is ringing [May 6 10:11:11] > 0x7f42d401c0f0 -- Strict RTP learning after remote address set to: 68.68.XX.XX:16862 [May 6 10:11:11] -- SIP/zsip-000000c2 is making progress passing it to Local/91601XXXXXXX@default-000000c5;2 [May 6 10:11:11] -- Local/91601XXXXXXX@default-000000c5;1 is making progress [May 6 10:11:34] -- SIP/zsip-000000c2 answered Local/91601XXXXXXX@default-000000c5;2 [May 6 10:11:34] -- Local/91601XXXXXXX@default-000000c5;1 answered [May 6 10:11:34] -- Channel SIP/zsip-000000c2 joined 'simple_bridge' basic-bridge <1fc0173b-9239-4a86-8c12-8b555e9e778e> [May 6 10:11:34] -- Executing [138368@default:1] AGI("Local/91601XXXXXXX@default-000000c5;1", "agi-VDAD_local_optimize.agi,V5061011050000000107") in new stack [May 6 10:11:34] == Manager 'sendcron' logged off from 127.0.0.1 [May 6 10:11:34] -- Channel Local/91601XXXXXXX@default-000000c5;2 joined 'simple_bridge' basic-bridge <1fc0173b-9239-4a86-8c12-8b555e9e778e> [May 6 10:11:34] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi [May 6 10:11:35] -- <Local/91601XXXXXXX@default-000000c5;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0 [May 6 10:11:35] -- Executing [138368@default:2] Wait("Local/91601XXXXXXX@default-000000c5;1", "2") in new stack [May 6 10:11:37] -- Executing [138368@default:3] Hangup("Local/91601XXXXXXX@default-000000c5;1", "") in new stack [May 6 10:11:37] == Spawn extension (default, 138368, 3) exited non-zero on 'Local/91601XXXXXXX@default-000000c5;1' [May 6 10:11:37] WARNING[19133][C-00000159]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [May 6 10:11:37] -- Executing [h@default:1] AGI("Local/91601XXXXXXX@default-000000c5;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [May 6 10:11:38] -- <Local/91601XXXXXXX@default-000000c5;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [May 6 10:11:38] -- Channel Local/91601XXXXXXX@default-000000c5;2 left 'simple_bridge' basic-bridge <1fc0173b-9239-4a86-8c12-8b555e9e778e> [May 6 10:11:38] -- Channel SIP/zsip-000000c2 left 'simple_bridge' basic-bridge <1fc0173b-9239-4a86-8c12-8b555e9e778e> [May 6 10:11:38] == Spawn extension (default, 91601XXXXXXX, 3) exited non-zero on 'Local/91601XXXXXXX@default-000000c5;2' [May 6 10:11:38] -- Executing [h@default:1] AGI("Local/91601XXXXXXX@default-000000c5;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----32-----3-----SIP 200 OK)") in new stack [May 6 10:11:38] -- <Local/91601XXXXXXX@default-000000c5;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----32-----3-----SIP 200 OK) completed, returning 0

I tried two different carriers but its happening on both of them. below is my carrier entry and dial plan.

[zsip]
disallow=all
allow=ulaw
allow=alaw
allow=g729
type=friend
host=69.xx.xx.xxx
port=5060
dtmfmode=rfc2833
qualify=yes
insecure=very
nat=force_rport,comedia
canreinvite=no

exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91X.,2,Set(CALLERID(ALL)="" <833xxxxxxx>)
exten => _91X.,3,Dial(SIP/1${EXTEN:2}@zsip,60,Tto)
exten => _91X.,4,Hangup

am i doing anything wrong? or is there any mistake i have made during installation and configuration?

ViciBox_v9.x86_64-9.0.2.ISO | Vicidial 2.14-751a Build 200425-0949 | Asterisk 13.29.2-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel(R) Xeon(R) CPU X5675 | 32 GB RAM | Load Average 0.13, 0.10, 0.09 |

Posted: **Sun May 10, 2020 12:38 pm**

Any Body ?

Posted: **Sun May 10, 2020 1:06 pm**

It appears everything is working the way its suppsoed to but just for shits and giggles, run this command and tell me the output:

dahdi_cfg -vvv

Posted: **Sun May 10, 2020 3:17 pm**

Here is the output from the command.

Code: Select all: DAHDI Tools Version - 2.11.1 DAHDI Version: 2.11.1 Echo Canceller(s): Configuration ====================== Channel map: 0 channels to configure.

Calls are not getting disconnected. it continues in the system until i disconnect them from the softphone.

there is a common warning in asterisk CLI i.e WARNING[17925][C-00000156]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel

this warning comes with every call I tested.

Posted: **Sun May 10, 2020 3:25 pm**

that is gonna happen, thats when you disconnect from the local channel, which is what dahdi provides which is why i said to run that command, but its running properly, honestly im stumped but perhaps make the dial time out longer may fix it.

So calls are being connected and you can speak to people its only when it "times out" ?

Posted: **Sun May 10, 2020 3:33 pm**

it seems to happen when it tries to spawn extension: 138368 so perhaps check the settings in extensions.conf for that particular extension and make sure it doesnt conflict with your dialplan....try adding a new phone and user and see if that works if u havent tried that already

Posted: **Tue May 12, 2020 2:06 pm**

Tried with the new user and new phone it's still the same issue, outbound autodialer calls hang up after a customer picks up the call. everything is working fine Inbound calls, outbound calls when dialed directly through phone extension. except for autodialer and the manual call which is dialed through agent interface. I tried dialing directly through the phone extension call connects fine.

Posted: **Tue May 12, 2020 2:20 pm**

1) SIP debug see what happens by comparing manual vs autodial. There's obviously a difference.

2) Did you upgrade your server at some point? Restore from a prior server and then upgrade the DB?

3) Please post the astersik version from /etc/astguiclient.conf and in admin->servers. Please also verify which version of sip.conf is in use. Does it match sip.conf.sample or sip.conf.sample-1.4?

Posted: **Wed May 13, 2020 12:42 pm**

Manual Dial

Code: Select all: [May 13 12:27:12] Connected to Asterisk 13.29.2-vici currently running on vicibox9 (pid = 2481) [May 13 12:28:04] --- (7 headers 0 lines) --- [May 13 12:28:04] Really destroying SIP dialog '76ca06ea5ff185a261b5556a0e21d7ed@192.xxx.x.xxx:5060' Method: OPTIONS [May 13 12:28:06] [May 13 12:28:06] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:06] [May 13 12:28:06] <-------------> [May 13 12:28:07] == Manager 'sendcron' logged on from 127.0.0.1 [May 13 12:28:07] == Manager 'sendcron' logged off from 127.0.0.1 [May 13 12:28:07] [May 13 12:28:07] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:07] INVITE sip:916015091286@192.xxx.x.xxx SIP/2.0 [May 13 12:28:07] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-9d3640455a4ad964-1--d87543-;rport [May 13 12:28:07] Max-Forwards: 70 [May 13 12:28:07] Contact: <sip:781@xxx.xxx.x.xxx:10219> [May 13 12:28:07] To: <sip:916015091286@192.xxx.x.xxx> [May 13 12:28:07] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:07] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:07] CSeq: 1 INVITE [May 13 12:28:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 13 12:28:07] Content-Type: application/sdp [May 13 12:28:07] Supported: eventlist [May 13 12:28:07] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:28:07] Content-Length: 244 [May 13 12:28:07] [May 13 12:28:07] v=0 [May 13 12:28:07] o=- 360955551 360955556 IN IP4 xxx.xxx.x.xxx [May 13 12:28:07] s=eyeBeam [May 13 12:28:07] c=IN IP4 xxx.xxx.x.xxx [May 13 12:28:07] t=0 0 [May 13 12:28:07] m=audio 8288 RTP/AVP 8 0 3 18 5 101 [May 13 12:28:07] a=alt:1 1 : 6F13C7D3 0000007D xxx.xxx.x.xxx 8288 [May 13 12:28:07] a=fmtp:101 0-15 [May 13 12:28:07] a=rtpmap:101 telephone-event/8000 [May 13 12:28:07] a=sendrecv [May 13 12:28:07] <-------------> [May 13 12:28:07] --- (13 headers 10 lines) --- [May 13 12:28:07] Sending to xxx.xxx.x.xxx:10219 (NAT) [May 13 12:28:07] Sending to xxx.xxx.x.xxx:10219 (NAT) [May 13 12:28:07] Using INVITE request as basis request - 762d8d7406168d4d@SkQtUEM. [May 13 12:28:07] Found peer '781' for '781' from xxx.xxx.x.xxx:10219 [May 13 12:28:07] [May 13 12:28:07] <--- Reliably Transmitting (NAT) to xxx.xxx.x.xxx:10219 ---> [May 13 12:28:07] SIP/2.0 401 Unauthorized [May 13 12:28:07] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-9d3640455a4ad964-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:07] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:07] To: <sip:916015091286@192.xxx.x.xxx>;tag=as7ede7ce2 [May 13 12:28:07] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:07] CSeq: 1 INVITE [May 13 12:28:07] Server: Asterisk PBX 13.29.2-vici [May 13 12:28:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:07] Supported: replaces, timer [May 13 12:28:07] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d1aa3dc" [May 13 12:28:07] Content-Length: 0 [May 13 12:28:07] [May 13 12:28:07] [May 13 12:28:07] <------------> [May 13 12:28:07] Scheduling destruction of SIP dialog '762d8d7406168d4d@SkQtUEM.' in 13376 ms (Method: INVITE) [May 13 12:28:07] Retransmitting #1 (NAT) to xxx.xxx.x.xxx:10219: [May 13 12:28:07] SIP/2.0 401 Unauthorized [May 13 12:28:07] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-9d3640455a4ad964-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:07] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:07] To: <sip:916015091286@192.xxx.x.xxx>;tag=as7ede7ce2 [May 13 12:28:07] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:07] CSeq: 1 INVITE [May 13 12:28:07] Server: Asterisk PBX 13.29.2-vici [May 13 12:28:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:07] Supported: replaces, timer [May 13 12:28:07] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d1aa3dc" [May 13 12:28:07] Content-Length: 0 [May 13 12:28:07] [May 13 12:28:07] [May 13 12:28:07] --- [May 13 12:28:07] [May 13 12:28:07] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:07] ACK sip:916015091286@192.xxx.x.xxx SIP/2.0 [May 13 12:28:07] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-9d3640455a4ad964-1--d87543-;rport [May 13 12:28:07] To: <sip:916015091286@192.xxx.x.xxx>;tag=as7ede7ce2 [May 13 12:28:07] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:07] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:07] CSeq: 1 ACK [May 13 12:28:07] Content-Length: 0 [May 13 12:28:07] [May 13 12:28:07] <-------------> [May 13 12:28:07] --- (7 headers 0 lines) --- [May 13 12:28:08] [May 13 12:28:08] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:08] INVITE sip:916015091286@192.xxx.x.xxx SIP/2.0 [May 13 12:28:08] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-2c465720b60c3448-1--d87543-;rport [May 13 12:28:08] Max-Forwards: 70 [May 13 12:28:08] Contact: <sip:781@xxx.xxx.x.xxx:10219> [May 13 12:28:08] To: <sip:916015091286@192.xxx.x.xxx> [May 13 12:28:08] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:08] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:08] CSeq: 2 INVITE [May 13 12:28:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 13 12:28:08] Content-Type: application/sdp [May 13 12:28:08] Supported: eventlist [May 13 12:28:08] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:28:08] Authorization: Digest username="781",realm="asterisk",nonce="6d1aa3dc",uri="sip:916015091286@192.xxx.x.xxx",response="69ca37308ff0e9ed3685730a3800634f",algorithm=MD5 [May 13 12:28:08] Content-Length: 244 [May 13 12:28:08] [May 13 12:28:08] v=0 [May 13 12:28:08] o=- 360955551 360955556 IN IP4 xxx.xxx.x.xxx [May 13 12:28:08] s=eyeBeam [May 13 12:28:08] c=IN IP4 xxx.xxx.x.xxx [May 13 12:28:08] t=0 0 [May 13 12:28:08] m=audio 8288 RTP/AVP 8 0 3 18 5 101 [May 13 12:28:08] a=alt:1 1 : 6F13C7D3 0000007D xxx.xxx.x.xxx 8288 [May 13 12:28:08] a=fmtp:101 0-15 [May 13 12:28:08] a=rtpmap:101 telephone-event/8000 [May 13 12:28:08] a=sendrecv [May 13 12:28:08] <-------------> [May 13 12:28:08] --- (14 headers 10 lines) --- [May 13 12:28:08] Sending to xxx.xxx.x.xxx:10219 (NAT) [May 13 12:28:08] Using INVITE request as basis request - 762d8d7406168d4d@SkQtUEM. [May 13 12:28:08] Found peer '781' for '781' from xxx.xxx.x.xxx:10219 [May 13 12:28:08] == Using SIP RTP CoS mark 5 [May 13 12:28:08] Found RTP audio format 8 [May 13 12:28:08] Found RTP audio format 0 [May 13 12:28:08] Found RTP audio format 3 [May 13 12:28:08] Found RTP audio format 18 [May 13 12:28:08] Found RTP audio format 5 [May 13 12:28:08] Found RTP audio format 101 [May 13 12:28:08] Found audio description format telephone-event for ID 101 [May 13 12:28:08] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|gsm|adpcm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|gsm) [May 13 12:28:08] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 13 12:28:08] > 0x7f5d9002c5f0 -- Strict RTP learning after remote address set to: xxx.xxx.x.xxx:8288 [May 13 12:28:08] Peer audio RTP is at port xxx.xxx.x.xxx:8288 [May 13 12:28:08] Looking for 916015091286 in default (domain 192.xxx.x.xxx) [May 13 12:28:08] sip_route_dump: route/path hop: <sip:781@xxx.xxx.x.xxx:10219> [May 13 12:28:08] [May 13 12:28:08] <--- Transmitting (NAT) to xxx.xxx.x.xxx:10219 ---> [May 13 12:28:08] SIP/2.0 100 Trying [May 13 12:28:08] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-2c465720b60c3448-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:08] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:08] To: <sip:916015091286@192.xxx.x.xxx> [May 13 12:28:08] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:08] CSeq: 2 INVITE [May 13 12:28:08] Server: Asterisk PBX 13.29.2-vici [May 13 12:28:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:08] Supported: replaces, timer [May 13 12:28:08] Contact: <sip:916015091286@192.xxx.x.xxx:5060> [May 13 12:28:08] Content-Length: 0 [May 13 12:28:08] [May 13 12:28:08] [May 13 12:28:08] <------------> [May 13 12:28:08] -- Executing [916015091286@default:1] AGI("SIP/781-00000112", "agi://127.0.0.1:4577/call_log") in new stack [May 13 12:28:08] -- <SIP/781-00000112>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [May 13 12:28:08] -- Executing [916015091286@default:2] Set("SIP/781-00000112", "CALLERID(ALL)="" <8336043243>") in new stack [May 13 12:28:08] -- Executing [916015091286@default:3] Dial("SIP/781-00000112", "SIP/16015091286@ahsan,60,Tto") in new stack [May 13 12:28:08] == Using SIP RTP CoS mark 5 [May 13 12:28:08] Audio is at 16390 [May 13 12:28:08] Adding codec ulaw to SDP [May 13 12:28:08] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:28:08] Reliably Transmitting (NAT) to 157.xxx.x.xxx:5060: [May 13 12:28:08] INVITE sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:28:08] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7b18a359;rport [May 13 12:28:08] Max-Forwards: 70 [May 13 12:28:08] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:08] To: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:08] Contact: <sip:XXXXX@192.xxx.x.xxx:5060> [May 13 12:28:08] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:08] CSeq: 102 INVITE [May 13 12:28:08] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:28:08] Date: Wed, 13 May 2020 16:28:08 GMT [May 13 12:28:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:08] Supported: replaces, timer [May 13 12:28:08] Remote-Party-ID: "8336043243" <sip:8336043243@192.xxx.x.xxx>;party=calling;privacy=off;screen=no [May 13 12:28:08] Content-Type: application/sdp [May 13 12:28:08] Content-Length: 259 [May 13 12:28:08] [May 13 12:28:08] v=0 [May 13 12:28:08] o=root 1034582360 1034582360 IN IP4 192.xxx.x.xxx [May 13 12:28:08] s=Asterisk PBX 13.29.2-vici [May 13 12:28:08] c=IN IP4 192.xxx.x.xxx [May 13 12:28:08] t=0 0 [May 13 12:28:08] m=audio 16390 RTP/AVP 0 101 [May 13 12:28:08] a=rtpmap:0 PCMU/8000 [May 13 12:28:08] a=rtpmap:101 telephone-event/8000 [May 13 12:28:08] a=fmtp:101 0-16 [May 13 12:28:08] a=ptime:20 [May 13 12:28:08] a=maxptime:150 [May 13 12:28:08] a=sendrecv [May 13 12:28:08] [May 13 12:28:08] --- [May 13 12:28:08] -- Called SIP/16015091286@ahsan [May 13 12:28:08] [May 13 12:28:08] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:08] ACK sip:916015091286@192.xxx.x.xxx SIP/2.0 [May 13 12:28:08] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-9d3640455a4ad964-1--d87543-;rport [May 13 12:28:08] To: <sip:916015091286@192.xxx.x.xxx>;tag=as7ede7ce2 [May 13 12:28:08] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:08] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:08] CSeq: 1 ACK [May 13 12:28:08] Content-Length: 0 [May 13 12:28:08] [May 13 12:28:08] <-------------> [May 13 12:28:08] --- (7 headers 0 lines) --- [May 13 12:28:08] [May 13 12:28:08] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:08] SIP/2.0 100 Trying [May 13 12:28:08] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7b18a359;received=103.8.112.182;rport=5060 [May 13 12:28:08] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:08] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:08] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:08] CSeq: 102 INVITE [May 13 12:28:08] Server: VOS3000 V2.1.4.0 [May 13 12:28:08] Content-Length: 0 [May 13 12:28:08] [May 13 12:28:08] <-------------> [May 13 12:28:08] --- (8 headers 0 lines) --- [May 13 12:28:08] [May 13 12:28:08] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:08] SIP/2.0 100 Trying [May 13 12:28:08] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7b18a359;received=103.8.112.182;rport=5060 [May 13 12:28:08] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:08] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:08] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:08] CSeq: 102 INVITE [May 13 12:28:08] Server: VOS3000 V2.1.4.0 [May 13 12:28:08] Content-Length: 0 [May 13 12:28:08] [May 13 12:28:08] <-------------> [May 13 12:28:08] --- (8 headers 0 lines) --- [May 13 12:28:09] [May 13 12:28:09] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:09] SIP/2.0 180 Ringing [May 13 12:28:09] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7b18a359;received=103.8.112.182;rport=5060 [May 13 12:28:09] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:09] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:09] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:09] CSeq: 102 INVITE [May 13 12:28:09] Contact: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:09] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK [May 13 12:28:09] Server: VOS3000 V2.1.4.0 [May 13 12:28:09] Content-Length: 0 [May 13 12:28:09] [May 13 12:28:09] <-------------> [May 13 12:28:09] --- (10 headers 0 lines) --- [May 13 12:28:09] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:09] -- SIP/ahsan-00000113 is ringing [May 13 12:28:09] [May 13 12:28:09] <--- Transmitting (NAT) to xxx.xxx.x.xxx:10219 ---> [May 13 12:28:09] SIP/2.0 180 Ringing [May 13 12:28:09] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-2c465720b60c3448-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:09] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:09] To: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:09] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:09] CSeq: 2 INVITE [May 13 12:28:09] Server: Asterisk PBX 13.29.2-vici [May 13 12:28:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:09] Supported: replaces, timer [May 13 12:28:09] Contact: <sip:916015091286@192.xxx.x.xxx:5060> [May 13 12:28:09] Content-Length: 0 [May 13 12:28:09] [May 13 12:28:09] [May 13 12:28:09] <------------> [May 13 12:28:10] [May 13 12:28:10] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:10] SIP/2.0 183 Session Progress [May 13 12:28:10] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7b18a359;received=103.8.112.182;rport=5060 [May 13 12:28:10] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:10] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:10] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:10] CSeq: 102 INVITE [May 13 12:28:10] Contact: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:10] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK [May 13 12:28:10] Server: VOS3000 V2.1.4.0 [May 13 12:28:10] Content-Type: application/sdp [May 13 12:28:10] Content-Length: 200 [May 13 12:28:10] [May 13 12:28:10] v=0 [May 13 12:28:10] o=- 8216 8216 IN IP4 157.xxx.x.xxx [May 13 12:28:10] s=VOS3000 [May 13 12:28:10] c=IN IP4 157.xxx.x.xxx [May 13 12:28:10] t=0 0 [May 13 12:28:10] m=audio 23726 RTP/AVP 0 101 [May 13 12:28:10] a=rtpmap:0 PCMU/8000 [May 13 12:28:10] a=rtpmap:101 telephone-event/8000 [May 13 12:28:10] a=fmtp:101 0-15 [May 13 12:28:10] a=sendrecv [May 13 12:28:10] <-------------> [May 13 12:28:10] --- (11 headers 10 lines) --- [May 13 12:28:10] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:10] Found RTP audio format 0 [May 13 12:28:10] Found RTP audio format 101 [May 13 12:28:10] Found audio description format PCMU for ID 0 [May 13 12:28:10] Found audio description format telephone-event for ID 101 [May 13 12:28:10] Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [May 13 12:28:10] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 13 12:28:10] > 0x7f5d4400c860 -- Strict RTP learning after remote address set to: 157.xxx.x.xxx:23726 [May 13 12:28:10] Peer audio RTP is at port 157.xxx.x.xxx:23726 [May 13 12:28:10] -- SIP/ahsan-00000113 is making progress passing it to SIP/781-00000112 [May 13 12:28:10] Audio is at 17712 [May 13 12:28:10] Adding codec ulaw to SDP [May 13 12:28:10] Adding codec gsm to SDP [May 13 12:28:10] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:28:10] [May 13 12:28:10] <--- Transmitting (NAT) to xxx.xxx.x.xxx:10219 ---> [May 13 12:28:10] SIP/2.0 183 Session Progress [May 13 12:28:10] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-2c465720b60c3448-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:10] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:10] To: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:10] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:10] CSeq: 2 INVITE [May 13 12:28:10] Server: Asterisk PBX 13.29.2-vici [May 13 12:28:10] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:10] Supported: replaces, timer [May 13 12:28:10] Contact: <sip:916015091286@192.xxx.x.xxx:5060> [May 13 12:28:10] Content-Type: application/sdp [May 13 12:28:10] Content-Length: 278 [May 13 12:28:10] [May 13 12:28:10] v=0 [May 13 12:28:10] o=root 76119618 76119618 IN IP4 192.xxx.x.xxx [May 13 12:28:10] s=Asterisk PBX 13.29.2-vici [May 13 12:28:10] c=IN IP4 192.xxx.x.xxx [May 13 12:28:10] t=0 0 [May 13 12:28:10] m=audio 17712 RTP/AVP 0 3 101 [May 13 12:28:10] a=rtpmap:0 PCMU/8000 [May 13 12:28:10] a=rtpmap:3 GSM/8000 [May 13 12:28:10] a=rtpmap:101 telephone-event/8000 [May 13 12:28:10] a=fmtp:101 0-16 [May 13 12:28:10] a=ptime:20 [May 13 12:28:10] a=maxptime:150 [May 13 12:28:10] a=sendrecv [May 13 12:28:10] [May 13 12:28:10] <------------> [May 13 12:28:10] > 0x7f5d9002c5f0 -- Strict RTP switching to RTP target address xxx.xxx.x.xxx:8288 as source [May 13 12:28:10] > 0x7f5d4400c860 -- Strict RTP switching to RTP target address 157.xxx.x.xxx:23726 as source [May 13 12:28:15] > 0x7f5d4400c860 -- Strict RTP learning complete - Locking on source address 157.xxx.x.xxx:23726 [May 13 12:28:15] [May 13 12:28:15] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:15] [May 13 12:28:15] <-------------> [May 13 12:28:23] [May 13 12:28:23] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:23] SIP/2.0 200 OK [May 13 12:28:23] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7b18a359;received=103.8.112.182;rport=5060 [May 13 12:28:23] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:23] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:23] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:23] CSeq: 102 INVITE [May 13 12:28:23] Contact: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:23] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK [May 13 12:28:23] Server: VOS3000 V2.1.4.0 [May 13 12:28:23] Supported: timer, linknat [May 13 12:28:23] Require: timer [May 13 12:28:23] Session-Expires: 600;refresher=uas [May 13 12:28:23] Content-Type: application/sdp [May 13 12:28:23] Content-Length: 200 [May 13 12:28:23] [May 13 12:28:23] v=0 [May 13 12:28:23] o=- 8216 8216 IN IP4 157.xxx.x.xxx [May 13 12:28:23] s=VOS3000 [May 13 12:28:23] c=IN IP4 157.xxx.x.xxx [May 13 12:28:23] t=0 0 [May 13 12:28:23] m=audio 23726 RTP/AVP 0 101 [May 13 12:28:23] a=rtpmap:0 PCMU/8000 [May 13 12:28:23] a=rtpmap:101 telephone-event/8000 [May 13 12:28:23] a=fmtp:101 0-15 [May 13 12:28:23] a=sendrecv [May 13 12:28:23] <-------------> [May 13 12:28:23] --- (14 headers 10 lines) --- [May 13 12:28:23] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:23] Transmitting (NAT) to 157.xxx.x.xxx:5060: [May 13 12:28:23] ACK sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:28:23] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7cdfcd21;rport [May 13 12:28:23] Max-Forwards: 70 [May 13 12:28:23] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:23] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:23] Contact: <sip:XXXXX@192.xxx.x.xxx:5060> [May 13 12:28:23] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:23] CSeq: 102 ACK [May 13 12:28:23] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:28:23] Content-Length: 0 [May 13 12:28:23] [May 13 12:28:23] [May 13 12:28:23] --- [May 13 12:28:23] -- SIP/ahsan-00000113 answered SIP/781-00000112 [May 13 12:28:23] Audio is at 17712 [May 13 12:28:23] Adding codec ulaw to SDP [May 13 12:28:23] Adding codec gsm to SDP [May 13 12:28:23] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:28:23] [May 13 12:28:23] <--- Reliably Transmitting (NAT) to xxx.xxx.x.xxx:10219 ---> [May 13 12:28:23] SIP/2.0 200 OK [May 13 12:28:23] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-2c465720b60c3448-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:23] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:23] To: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:23] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:23] CSeq: 2 INVITE [May 13 12:28:23] Server: Asterisk PBX 13.29.2-vici [May 13 12:28:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:23] Supported: replaces, timer [May 13 12:28:23] Contact: <sip:916015091286@192.xxx.x.xxx:5060> [May 13 12:28:23] Content-Type: application/sdp [May 13 12:28:23] Content-Length: 278 [May 13 12:28:23] [May 13 12:28:23] v=0 [May 13 12:28:23] o=root 76119618 76119618 IN IP4 192.xxx.x.xxx [May 13 12:28:23] s=Asterisk PBX 13.29.2-vici [May 13 12:28:23] c=IN IP4 192.xxx.x.xxx [May 13 12:28:23] t=0 0 [May 13 12:28:23] m=audio 17712 RTP/AVP 0 3 101 [May 13 12:28:23] a=rtpmap:0 PCMU/8000 [May 13 12:28:23] a=rtpmap:3 GSM/8000 [May 13 12:28:23] a=rtpmap:101 telephone-event/8000 [May 13 12:28:23] a=fmtp:101 0-16 [May 13 12:28:23] a=ptime:20 [May 13 12:28:23] a=maxptime:150 [May 13 12:28:23] a=sendrecv [May 13 12:28:23] [May 13 12:28:23] <------------> [May 13 12:28:23] -- Channel SIP/ahsan-00000113 joined 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:23] -- Channel SIP/781-00000112 joined 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:23] > 0x7f5d9002c5f0 -- Strict RTP learning complete - Locking on source address xxx.xxx.x.xxx:8288 [May 13 12:28:23] [May 13 12:28:23] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:23] ACK sip:916015091286@192.xxx.x.xxx:5060 SIP/2.0 [May 13 12:28:23] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-f66d0e26484a8263-1--d87543-;rport [May 13 12:28:23] Max-Forwards: 70 [May 13 12:28:23] Contact: <sip:781@xxx.xxx.x.xxx:10219> [May 13 12:28:23] To: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:23] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:23] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:23] CSeq: 2 ACK [May 13 12:28:23] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:28:23] Content-Length: 0 [May 13 12:28:23] [May 13 12:28:23] <-------------> [May 13 12:28:23] --- (10 headers 0 lines) --- [May 13 12:28:24] [May 13 12:28:24] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:24] [May 13 12:28:24] <-------------> [May 13 12:28:31] Reliably Transmitting (NAT) to xxx.xxx.x.xxx:10219: [May 13 12:28:31] OPTIONS sip:781@xxx.xxx.x.xxx:10219 SIP/2.0 [May 13 12:28:31] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK2eb40e83;rport [May 13 12:28:31] Max-Forwards: 70 [May 13 12:28:31] From: "asterisk" <sip:asterisk@192.xxx.x.xxx>;tag=as5d0d4c20 [May 13 12:28:31] To: <sip:781@xxx.xxx.x.xxx:10219> [May 13 12:28:31] Contact: <sip:asterisk@192.xxx.x.xxx:5060> [May 13 12:28:31] Call-ID: 17edc0194c11fe48254734c95200541c@192.xxx.x.xxx:5060 [May 13 12:28:31] CSeq: 102 OPTIONS [May 13 12:28:31] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:28:31] Date: Wed, 13 May 2020 16:28:31 GMT [May 13 12:28:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:31] Supported: replaces, timer [May 13 12:28:31] Content-Length: 0 [May 13 12:28:31] [May 13 12:28:31] [May 13 12:28:31] --- [May 13 12:28:31] [May 13 12:28:31] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:31] SIP/2.0 200 OK [May 13 12:28:31] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK2eb40e83;rport=5060 [May 13 12:28:31] Contact: <sip:xxx.xxx.x.xxx:10219> [May 13 12:28:31] To: <sip:781@xxx.xxx.x.xxx:10219>;tag=6b76b048 [May 13 12:28:31] From: "asterisk"<sip:asterisk@192.xxx.x.xxx>;tag=as5d0d4c20 [May 13 12:28:31] Call-ID: 17edc0194c11fe48254734c95200541c@192.xxx.x.xxx:5060 [May 13 12:28:31] CSeq: 102 OPTIONS [May 13 12:28:31] Accept: application/sdp [May 13 12:28:31] Accept-Language: en [May 13 12:28:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 13 12:28:31] Supported: eventlist [May 13 12:28:31] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:28:31] Content-Length: 0 [May 13 12:28:31] [May 13 12:28:31] <-------------> [May 13 12:28:31] --- (13 headers 0 lines) --- [May 13 12:28:31] Really destroying SIP dialog '17edc0194c11fe48254734c95200541c@192.xxx.x.xxx:5060' Method: OPTIONS [May 13 12:28:33] [May 13 12:28:33] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:33] [May 13 12:28:33] <-------------> [May 13 12:28:34] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:34] SIP/2.0 200 OK [May 13 12:28:34] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK318ce6b8;received=103.8.112.182;rport=5060 [May 13 12:28:34] From: <sip:XXXXX@157.xxx.x.xxx>;tag=as070a8fa3 [May 13 12:28:34] To: <sip:XXXXX@157.xxx.x.xxx> [May 13 12:28:34] Call-ID: 6a07608e773eb2223f5417d4624abd15@192.xxx.x.xxx [May 13 12:28:34] CSeq: 105 REGISTER [May 13 12:28:34] Content-Length: 0 [May 13 12:28:34] Contact: <sip:s@192.xxx.x.xxx:5060> [May 13 12:28:34] Expires: 60 [May 13 12:28:34] [May 13 12:28:34] <-------------> [May 13 12:28:34] --- (9 headers 0 lines) --- [May 13 12:28:34] NOTICE[2594]: chan_sip.c:24624 handle_response_register: Outbound Registration: Expiry for 157.xxx.x.xxx is 60 sec (Scheduling reregistration in 45 s) [May 13 12:28:34] Really destroying SIP dialog '6a07608e773eb2223f5417d4624abd15@192.xxx.x.xxx' Method: REGISTER [May 13 12:28:35] [May 13 12:28:35] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:35] BYE sip:XXXXX@192.xxx.x.xxx:5060 SIP/2.0 [May 13 12:28:35] Via: SIP/2.0/UDP 157.xxx.x.xxx:5060;branch=z9hG4bK040135a26142b78e [May 13 12:28:35] From: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:35] To: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:35] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:35] CSeq: 2140 BYE [May 13 12:28:35] Max-Forwards: 70 [May 13 12:28:35] Content-Length: 0 [May 13 12:28:35] [May 13 12:28:35] <-------------> [May 13 12:28:35] --- (8 headers 0 lines) --- [May 13 12:28:35] Sending to 157.xxx.x.xxx:5060 (NAT) [May 13 12:28:35] Scheduling destruction of SIP dialog '3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060' in 6400 ms (Method: BYE) [May 13 12:28:35] [May 13 12:28:35] <--- Transmitting (NAT) to 157.xxx.x.xxx:5060 ---> [May 13 12:28:35] SIP/2.0 200 OK [May 13 12:28:35] Via: SIP/2.0/UDP 157.xxx.x.xxx:5060;branch=z9hG4bK040135a26142b78e;received=157.xxx.x.xxx;rport=5060 [May 13 12:28:35] From: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:35] To: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:35] Call-ID: 3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060 [May 13 12:28:35] CSeq: 2140 BYE [May 13 12:28:35] Server: Asterisk PBX 13.29.2-vici [May 13 12:28:35] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:28:35] Supported: replaces, timer [May 13 12:28:35] Content-Length: 0 [May 13 12:28:35] [May 13 12:28:35] [May 13 12:28:35] <------------> [May 13 12:28:35] -- Channel SIP/ahsan-00000113 left 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:35] -- Channel SIP/781-00000112 left 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:35] == Spawn extension (default, 916015091286, 3) exited non-zero on 'SIP/781-00000112' [May 13 12:28:35] -- Executing [h@default:1] AGI("SIP/781-00000112", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----27-----11-----SIP 200 OK)") in new stack [May 13 12:28:35] -- <SIP/781-00000112>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----27-----11-----SIP 200 OK) completed, returning 0 [May 13 12:28:35] Scheduling destruction of SIP dialog '762d8d7406168d4d@SkQtUEM.' in 13376 ms (Method: ACK) [May 13 12:28:35] Reliably Transmitting (NAT) to xxx.xxx.x.xxx:10219: [May 13 12:28:35] BYE sip:781@xxx.xxx.x.xxx:10219 SIP/2.0 [May 13 12:28:35] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK65101929;rport [May 13 12:28:35] Max-Forwards: 70 [May 13 12:28:35] From: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:35] To: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:35] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:35] CSeq: 102 BYE [May 13 12:28:35] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:28:35] Proxy-Authorization: Digest username="781", realm="asterisk", algorithm=MD5, uri="sip:192.xxx.x.xxx", nonce="6d1aa3dc", response="a491dc8e8c77f25a22f946bef0a52be6" [May 13 12:28:35] X-Asterisk-HangupCause: Normal Clearing [May 13 12:28:35] X-Asterisk-HangupCauseCode: 16 [May 13 12:28:35] Content-Length: 0

Audialer Call

Code: Select all: [May 13 12:30:13] == Manager 'sendcron' logged on from 127.0.0.1 [May 13 12:30:13] == Using SIP RTP CoS mark 5 [May 13 12:30:13] Audio is at 18298 [May 13 12:30:13] Adding codec ulaw to SDP [May 13 12:30:13] Adding codec gsm to SDP [May 13 12:30:13] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:30:13] Reliably Transmitting (NAT) to 192.xxx.x.xxx:10219: [May 13 12:30:13] INVITE sip:781@192.xxx.x.xxx:10219 SIP/2.0 [May 13 12:30:13] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK63245d7b;rport [May 13 12:30:13] Max-Forwards: 70 [May 13 12:30:13] From: "S2005131230138600051" <sip:855xxxxxxx@192.xxx.x.xxx>;tag=as679cf9b6 [May 13 12:30:13] To: <sip:781@192.xxx.x.xxx:10219> [May 13 12:30:13] Contact: <sip:855xxxxxxx@192.xxx.x.xxx:5060> [May 13 12:30:13] Call-ID: 0426e4ca0b0e4ce9171b26e976c14b11@192.xxx.x.xxx:5060 [May 13 12:30:13] CSeq: 102 INVITE [May 13 12:30:13] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:30:13] Date: Wed, 13 May 2020 16:30:13 GMT [May 13 12:30:13] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:30:13] Supported: replaces, timer [May 13 12:30:13] Remote-Party-ID: "S2005131230138600051" <sip:855xxxxxxx@192.xxx.x.xxx>;party=calling;privacy=off;screen=no [May 13 12:30:13] Content-Type: application/sdp [May 13 12:30:13] Content-Length: 280 [May 13 12:30:13] [May 13 12:30:13] v=0 [May 13 12:30:13] o=root 882267685 882267685 IN IP4 192.xxx.x.xxx [May 13 12:30:13] s=Asterisk PBX 13.29.2-vici [May 13 12:30:13] c=IN IP4 192.xxx.x.xxx [May 13 12:30:13] t=0 0 [May 13 12:30:13] m=audio 18298 RTP/AVP 0 3 101 [May 13 12:30:13] a=rtpmap:0 PCMU/8000 [May 13 12:30:13] a=rtpmap:3 GSM/8000 [May 13 12:30:13] a=rtpmap:101 telephone-event/8000 [May 13 12:30:13] a=fmtp:101 0-16 [May 13 12:30:13] a=ptime:20 [May 13 12:30:13] a=maxptime:150 [May 13 12:30:13] a=sendrecv [May 13 12:30:13] [May 13 12:30:13] --- [May 13 12:30:13] -- Called 781 [May 13 12:30:13] [May 13 12:30:13] <--- SIP read from UDP:192.xxx.x.xxx:10219 ---> [May 13 12:30:13] SIP/2.0 180 Ringing [May 13 12:30:13] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK63245d7b;rport=5060 [May 13 12:30:13] Contact: <sip:781@192.xxx.x.xxx:10219> [May 13 12:30:13] To: <sip:781@192.xxx.x.xxx:10219>;tag=45268f3d [May 13 12:30:13] From: "S2005131230138600051"<sip:855xxxxxxx@192.xxx.x.xxx>;tag=as679cf9b6 [May 13 12:30:13] Call-ID: 0426e4ca0b0e4ce9171b26e976c14b11@192.xxx.x.xxx:5060 [May 13 12:30:13] CSeq: 102 INVITE [May 13 12:30:13] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:30:13] Content-Length: 0 [May 13 12:30:13] [May 13 12:30:13] <-------------> [May 13 12:30:13] --- (9 headers 0 lines) --- [May 13 12:30:13] sip_route_dump: route/path hop: <sip:781@192.xxx.x.xxx:10219> [May 13 12:30:13] -- SIP/781-00000114 is ringing [May 13 12:30:15] [May 13 12:30:15] <--- SIP read from UDP:192.xxx.x.xxx:10219 ---> [May 13 12:30:15] SIP/2.0 200 OK [May 13 12:30:15] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK63245d7b;rport=5060 [May 13 12:30:15] Contact: <sip:781@192.xxx.x.xxx:10219> [May 13 12:30:15] To: <sip:781@192.xxx.x.xxx:10219>;tag=45268f3d [May 13 12:30:15] From: "S2005131230138600051"<sip:855xxxxxxx@192.xxx.x.xxx>;tag=as679cf9b6 [May 13 12:30:15] Call-ID: 0426e4ca0b0e4ce9171b26e976c14b11@192.xxx.x.xxx:5060 [May 13 12:30:15] CSeq: 102 INVITE [May 13 12:30:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 13 12:30:15] Content-Type: application/sdp [May 13 12:30:15] Supported: eventlist [May 13 12:30:15] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:30:15] Content-Length: 237 [May 13 12:30:15] [May 13 12:30:15] v=0 [May 13 12:30:15] o=- 361081374 361081380 IN IP4 192.xxx.x.xxx [May 13 12:30:15] s=eyeBeam [May 13 12:30:15] c=IN IP4 192.xxx.x.xxx [May 13 12:30:15] t=0 0 [May 13 12:30:15] m=audio 8288 RTP/AVP 0 3 101 [May 13 12:30:15] a=alt:1 1 : 01A4DA61 00000007 192.xxx.x.xxx 8288 [May 13 12:30:15] a=fmtp:101 0-15 [May 13 12:30:15] a=rtpmap:101 telephone-event/8000 [May 13 12:30:15] a=sendrecv [May 13 12:30:15] <-------------> [May 13 12:30:15] --- (12 headers 10 lines) --- [May 13 12:30:15] Found RTP audio format 0 [May 13 12:30:15] Found RTP audio format 3 [May 13 12:30:15] Found RTP audio format 101 [May 13 12:30:15] Found audio description format telephone-event for ID 101 [May 13 12:30:15] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|gsm)/video=(nothing)/text=(nothing), combined - (ulaw|gsm) [May 13 12:30:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 13 12:30:15] > 0x7f5d3800fc00 -- Strict RTP learning after remote address set to: 192.xxx.x.xxx:8288 [May 13 12:30:15] Peer audio RTP is at port 192.xxx.x.xxx:8288 [May 13 12:30:15] sip_route_dump: route/path hop: <sip:781@192.xxx.x.xxx:10219> [May 13 12:30:15] Transmitting (NAT) to 192.xxx.x.xxx:10219: [May 13 12:30:15] ACK sip:781@192.xxx.x.xxx:10219 SIP/2.0 [May 13 12:30:15] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK76062a5d;rport [May 13 12:30:15] Max-Forwards: 70 [May 13 12:30:15] From: "S2005131230138600051" <sip:855xxxxxxx@192.xxx.x.xxx>;tag=as679cf9b6 [May 13 12:30:15] To: <sip:781@192.xxx.x.xxx:10219>;tag=45268f3d [May 13 12:30:15] Contact: <sip:855xxxxxxx@192.xxx.x.xxx:5060> [May 13 12:30:15] Call-ID: 0426e4ca0b0e4ce9171b26e976c14b11@192.xxx.x.xxx:5060 [May 13 12:30:15] CSeq: 102 ACK [May 13 12:30:15] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:30:15] Content-Length: 0 [May 13 12:30:15] [May 13 12:30:15] [May 13 12:30:15] --- [May 13 12:30:15] -- SIP/781-00000114 answered [May 13 12:30:15] -- Executing [8600051@default:1] MeetMe("SIP/781-00000114", "8600051,F") in new stack [May 13 12:30:15] -- Created MeetMe conference 1023 for conference '8600051' [May 13 12:30:15] -- <SIP/781-00000114> Playing 'conf-onlyperson.slin' (language 'en') [May 13 12:30:15] > 0x7f5d3800fc00 -- Strict RTP switching to RTP target address 192.xxx.x.xxx:8288 as source [May 13 12:30:16] == Manager 'sendcron' logged off from 127.0.0.1 [May 13 12:30:22] [May 13 12:30:22] <--- SIP read from UDP:192.xxx.x.xxx:10219 ---> [May 13 12:30:22] [May 13 12:30:22] <-------------> [May 13 12:30:22] == Manager 'sendcron' logged on from 127.0.0.1 [May 13 12:30:22] -- Called 916015091286@default [May 13 12:30:22] -- Executing [916015091286@default:1] AGI("Local/916015091286@default-0000001d;2", "agi://127.0.0.1:4577/call_log") in new stack [May 13 12:30:22] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=001)) [May 13 12:30:22] -- <Local/916015091286@default-0000001d;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [May 13 12:30:22] -- Executing [916015091286@default:2] Set("Local/916015091286@default-0000001d;2", "CALLERID(ALL)="" <8336043243>") in new stack [May 13 12:30:22] -- Executing [916015091286@default:3] Dial("Local/916015091286@default-0000001d;2", "SIP/16015091286@ahsan,60,Tto") in new stack [May 13 12:30:22] == Using SIP RTP CoS mark 5 [May 13 12:30:22] Audio is at 13352 [May 13 12:30:22] Adding codec ulaw to SDP [May 13 12:30:22] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:30:22] Reliably Transmitting (NAT) to 157.xxx.x.xxx:5060: [May 13 12:30:22] INVITE sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:30:22] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;rport [May 13 12:30:22] Max-Forwards: 70 [May 13 12:30:22] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:22] To: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:22] Contact: <sip:XXXXX@192.xxx.x.xxx:5060> [May 13 12:30:22] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:22] CSeq: 102 INVITE [May 13 12:30:22] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:30:22] Date: Wed, 13 May 2020 16:30:22 GMT [May 13 12:30:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:30:22] Supported: replaces, timer [May 13 12:30:22] Remote-Party-ID: "8336043243" <sip:8336043243@192.xxx.x.xxx>;party=calling;privacy=off;screen=no [May 13 12:30:22] Content-Type: application/sdp [May 13 12:30:22] Content-Length: 259 [May 13 12:30:22] [May 13 12:30:22] v=0 [May 13 12:30:22] o=root 2043095914 2043095914 IN IP4 192.xxx.x.xxx [May 13 12:30:22] s=Asterisk PBX 13.29.2-vici [May 13 12:30:22] c=IN IP4 192.xxx.x.xxx [May 13 12:30:22] t=0 0 [May 13 12:30:22] m=audio 13352 RTP/AVP 0 101 [May 13 12:30:22] a=rtpmap:0 PCMU/8000 [May 13 12:30:22] a=rtpmap:101 telephone-event/8000 [May 13 12:30:22] a=fmtp:101 0-16 [May 13 12:30:22] a=ptime:20 [May 13 12:30:22] a=maxptime:150 [May 13 12:30:22] a=sendrecv [May 13 12:30:22] [May 13 12:30:22] --- [May 13 12:30:22] -- Called SIP/16015091286@ahsan [May 13 12:30:22] Retransmitting #1 (NAT) to 157.xxx.x.xxx:5060: [May 13 12:30:22] INVITE sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:30:22] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;rport [May 13 12:30:22] Max-Forwards: 70 [May 13 12:30:22] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:22] To: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:22] Contact: <sip:XXXXX@192.xxx.x.xxx:5060> [May 13 12:30:22] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:22] CSeq: 102 INVITE [May 13 12:30:22] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:30:22] Date: Wed, 13 May 2020 16:30:22 GMT [May 13 12:30:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:30:22] Supported: replaces, timer [May 13 12:30:22] Remote-Party-ID: "8336043243" <sip:8336043243@192.xxx.x.xxx>;party=calling;privacy=off;screen=no [May 13 12:30:22] Content-Type: application/sdp [May 13 12:30:22] Content-Length: 259 [May 13 12:30:22] [May 13 12:30:22] v=0 [May 13 12:30:22] o=root 2043095914 2043095914 IN IP4 192.xxx.x.xxx [May 13 12:30:22] s=Asterisk PBX 13.29.2-vici [May 13 12:30:22] c=IN IP4 192.xxx.x.xxx [May 13 12:30:22] t=0 0 [May 13 12:30:22] m=audio 13352 RTP/AVP 0 101 [May 13 12:30:22] a=rtpmap:0 PCMU/8000 [May 13 12:30:22] a=rtpmap:101 telephone-event/8000 [May 13 12:30:22] a=fmtp:101 0-16 [May 13 12:30:22] a=ptime:20 [May 13 12:30:22] a=maxptime:150 [May 13 12:30:22] a=sendrecv [May 13 12:30:22] [May 13 12:30:22] --- [May 13 12:30:22] [May 13 12:30:22] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:30:22] SIP/2.0 100 Trying [May 13 12:30:22] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;received=103.8.112.182;rport=5060 [May 13 12:30:22] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:22] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:22] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:22] CSeq: 102 INVITE [May 13 12:30:22] Server: VOS3000 V2.1.4.0 [May 13 12:30:22] Content-Length: 0 [May 13 12:30:22] [May 13 12:30:22] <-------------> [May 13 12:30:22] --- (8 headers 0 lines) --- [May 13 12:30:22] [May 13 12:30:22] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:30:22] SIP/2.0 100 Trying [May 13 12:30:22] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;received=103.8.112.182;rport=5060 [May 13 12:30:22] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:22] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:22] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:22] CSeq: 102 INVITE [May 13 12:30:22] Server: VOS3000 V2.1.4.0 [May 13 12:30:22] Content-Length: 0 [May 13 12:30:22] [May 13 12:30:22] <-------------> [May 13 12:30:22] --- (8 headers 0 lines) --- [May 13 12:30:22] [May 13 12:30:22] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:30:22] SIP/2.0 100 Trying [May 13 12:30:22] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;received=103.8.112.182;rport=5060 [May 13 12:30:22] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:22] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:22] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:22] CSeq: 102 INVITE [May 13 12:30:22] Server: VOS3000 V2.1.4.0 [May 13 12:30:22] Content-Length: 0 [May 13 12:30:22] [May 13 12:30:22] <-------------> [May 13 12:30:22] --- (8 headers 0 lines) --- [May 13 12:30:23] [May 13 12:30:23] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:30:23] SIP/2.0 180 Ringing [May 13 12:30:23] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;received=103.8.112.182;rport=5060 [May 13 12:30:23] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:23] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:23] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:23] CSeq: 102 INVITE [May 13 12:30:23] Contact: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:23] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK [May 13 12:30:23] Server: VOS3000 V2.1.4.0 [May 13 12:30:23] Content-Length: 0 [May 13 12:30:23] [May 13 12:30:23] <-------------> [May 13 12:30:23] --- (10 headers 0 lines) --- [May 13 12:30:23] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:23] -- SIP/ahsan-00000115 is ringing [May 13 12:30:23] -- Local/916015091286@default-0000001d;1 is ringing [May 13 12:30:24] [May 13 12:30:24] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:30:24] SIP/2.0 183 Session Progress [May 13 12:30:24] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;received=103.8.112.182;rport=5060 [May 13 12:30:24] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:24] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:24] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:24] CSeq: 102 INVITE [May 13 12:30:24] Contact: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:24] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK [May 13 12:30:24] Server: VOS3000 V2.1.4.0 [May 13 12:30:24] Content-Type: application/sdp [May 13 12:30:24] Content-Length: 200 [May 13 12:30:24] [May 13 12:30:24] v=0 [May 13 12:30:24] o=- 8350 8350 IN IP4 157.xxx.x.xxx [May 13 12:30:24] s=VOS3000 [May 13 12:30:24] c=IN IP4 157.xxx.x.xxx [May 13 12:30:24] t=0 0 [May 13 12:30:24] m=audio 24638 RTP/AVP 0 101 [May 13 12:30:24] a=rtpmap:0 PCMU/8000 [May 13 12:30:24] a=rtpmap:101 telephone-event/8000 [May 13 12:30:24] a=fmtp:101 0-15 [May 13 12:30:24] a=sendrecv [May 13 12:30:24] <-------------> [May 13 12:30:24] --- (11 headers 10 lines) --- [May 13 12:30:24] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:24] Found RTP audio format 0 [May 13 12:30:24] Found RTP audio format 101 [May 13 12:30:24] Found audio description format PCMU for ID 0 [May 13 12:30:24] Found audio description format telephone-event for ID 101 [May 13 12:30:24] Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [May 13 12:30:24] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 13 12:30:24] > 0x7f5d4000bc50 -- Strict RTP learning after remote address set to: 157.xxx.x.xxx:24638 [May 13 12:30:24] Peer audio RTP is at port 157.xxx.x.xxx:24638 [May 13 12:30:24] -- SIP/ahsan-00000115 is making progress passing it to Local/916015091286@default-0000001d;2 [May 13 12:30:24] -- Local/916015091286@default-0000001d;1 is making progress [May 13 12:30:31] [May 13 12:30:31] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:30:31] SIP/2.0 200 OK [May 13 12:30:31] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK17cc7878;received=103.8.112.182;rport=5060 [May 13 12:30:31] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:31] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:31] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:31] CSeq: 102 INVITE [May 13 12:30:31] Contact: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:31] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK [May 13 12:30:31] Server: VOS3000 V2.1.4.0 [May 13 12:30:31] Supported: timer, linknat [May 13 12:30:31] Require: timer [May 13 12:30:31] Session-Expires: 600;refresher=uas [May 13 12:30:31] Content-Type: application/sdp [May 13 12:30:31] Content-Length: 200 [May 13 12:30:31] [May 13 12:30:31] v=0 [May 13 12:30:31] o=- 8350 8350 IN IP4 157.xxx.x.xxx [May 13 12:30:31] s=VOS3000 [May 13 12:30:31] c=IN IP4 157.xxx.x.xxx [May 13 12:30:31] t=0 0 [May 13 12:30:31] m=audio 24638 RTP/AVP 0 101 [May 13 12:30:31] a=rtpmap:0 PCMU/8000 [May 13 12:30:31] a=rtpmap:101 telephone-event/8000 [May 13 12:30:31] a=fmtp:101 0-15 [May 13 12:30:31] a=sendrecv [May 13 12:30:31] <-------------> [May 13 12:30:31] --- (14 headers 10 lines) --- [May 13 12:30:31] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:30:31] Transmitting (NAT) to 157.xxx.x.xxx:5060: [May 13 12:30:31] ACK sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:30:31] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK479994a5;rport [May 13 12:30:31] Max-Forwards: 70 [May 13 12:30:31] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:31] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:31] Contact: <sip:XXXXX@192.xxx.x.xxx:5060> [May 13 12:30:31] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:31] CSeq: 102 ACK [May 13 12:30:31] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:30:31] Content-Length: 0 [May 13 12:30:31] [May 13 12:30:31] [May 13 12:30:31] --- [May 13 12:30:31] -- SIP/ahsan-00000115 answered Local/916015091286@default-0000001d;2 [May 13 12:30:31] -- Local/916015091286@default-0000001d;1 answered [May 13 12:30:31] -- Executing [138369@default:1] AGI("Local/916015091286@default-0000001d;1", "agi-VDAD_local_optimize.agi,V5131230220000000208") in new stack [May 13 12:30:31] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi [May 13 12:30:31] -- Channel SIP/ahsan-00000115 joined 'simple_bridge' basic-bridge <6e2b3092-304c-4191-bbc3-f6b6f81f6962> [May 13 12:30:31] -- Channel Local/916015091286@default-0000001d;2 joined 'simple_bridge' basic-bridge <6e2b3092-304c-4191-bbc3-f6b6f81f6962> [May 13 12:30:31] [May 13 12:30:31] <--- SIP read from UDP:192.xxx.x.xxx:10219 ---> [May 13 12:30:31] SIP/2.0 200 OK [May 13 12:30:31] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK085e85e9;rport=5060 [May 13 12:30:31] Contact: <sip:192.xxx.x.xxx:10219> [May 13 12:30:31] To: <sip:781@192.xxx.x.xxx:10219>;tag=7f2a3b70 [May 13 12:30:31] From: "asterisk"<sip:asterisk@192.xxx.x.xxx>;tag=as5f64eac7 [May 13 12:30:31] Call-ID: 1dd59f1702f9f1a67c1d72f846e56e20@192.xxx.x.xxx:5060 [May 13 12:30:31] CSeq: 102 OPTIONS [May 13 12:30:31] Accept: application/sdp [May 13 12:30:31] Accept-Language: en [May 13 12:30:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 13 12:30:31] Supported: eventlist [May 13 12:30:31] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:30:31] Content-Length: 0 [May 13 12:30:31] [May 13 12:30:31] <-------------> [May 13 12:30:31] --- (13 headers 0 lines) --- [May 13 12:30:31] Really destroying SIP dialog '1dd59f1702f9f1a67c1d72f846e56e20@192.xxx.x.xxx:5060' Method: OPTIONS [May 13 12:30:31] -- <Local/916015091286@default-0000001d;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0 [May 13 12:30:31] -- Executing [138369@default:2] Wait("Local/916015091286@default-0000001d;1", "2") in new stack [May 13 12:30:32] == Manager 'sendcron' logged off from 127.0.0.1 [May 13 12:30:33] -- Executing [138369@default:3] Hangup("Local/916015091286@default-0000001d;1", "") in new stack [May 13 12:30:33] == Spawn extension (default, 138369, 3) exited non-zero on 'Local/916015091286@default-0000001d;1' [May 13 12:30:33] WARNING[18483][C-00000116]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [May 13 12:30:33] -- Executing [h@default:1] AGI("Local/916015091286@default-0000001d;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [May 13 12:30:34] -- <Local/916015091286@default-0000001d;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [May 13 12:30:34] -- Channel Local/916015091286@default-0000001d;2 left 'simple_bridge' basic-bridge <6e2b3092-304c-4191-bbc3-f6b6f81f6962> [May 13 12:30:34] -- Channel SIP/ahsan-00000115 left 'simple_bridge' basic-bridge <6e2b3092-304c-4191-bbc3-f6b6f81f6962> [May 13 12:30:34] == Spawn extension (default, 916015091286, 3) exited non-zero on 'Local/916015091286@default-0000001d;2' [May 13 12:30:34] Scheduling destruction of SIP dialog '1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060' in 6528 ms (Method: INVITE) [May 13 12:30:34] -- Executing [h@default:1] AGI("Local/916015091286@default-0000001d;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----3-----SIP 200 OK)") in new stack [May 13 12:30:34] Reliably Transmitting (NAT) to 157.xxx.x.xxx:5060: [May 13 12:30:34] BYE sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:30:34] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK3bd40bdd;rport [May 13 12:30:34] Max-Forwards: 70 [May 13 12:30:34] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:34] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:34] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:34] CSeq: 103 BYE [May 13 12:30:34] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:30:34] X-Asterisk-HangupCause: Normal Clearing [May 13 12:30:34] X-Asterisk-HangupCauseCode: 16 [May 13 12:30:34] Content-Length: 0 [May 13 12:30:34] [May 13 12:30:34] [May 13 12:30:34] --- [May 13 12:30:34] -- <Local/916015091286@default-0000001d;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----3-----SIP 200 OK) completed, returning 0 [May 13 12:30:34] [May 13 12:30:34] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:30:34] SIP/2.0 200 OK [May 13 12:30:34] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK3bd40bdd;received=103.8.112.182;rport=5060 [May 13 12:30:34] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as3c1c8a81 [May 13 12:30:34] To: <sip:16015091286@157.xxx.x.xxx:5060>;tag=00568e891bbdb093 [May 13 12:30:34] Call-ID: 1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060 [May 13 12:30:34] CSeq: 103 BYE [May 13 12:30:34] Content-Length: 0 [May 13 12:30:34] [May 13 12:30:34] <-------------> [May 13 12:30:34] --- (7 headers 0 lines) --- [May 13 12:30:34] Really destroying SIP dialog '1b5417c211966a340e2a58de77678c86@192.xxx.x.xxx:5060' Method: INVITE [May 13 12:30:36] > 0x7f5d3800fc00 -- Strict RTP learning complete - Locking on source address 192.xxx.x.xxx:8288 [May 13 12:30:40] [May 13 12:30:40] <--- SIP read from UDP:192.xxx.x.xxx:10219 ---> [May 13 12:30:40] [May 13 12:30:40] <-------------> [May 13 12:30:49] [May 13 12:30:49] <--- SIP read from UDP:192.xxx.x.xxx:10219 ---> [May 13 12:30:49] [May 13 12:30:49] <-------------> [May 13 12:31:34] --- (9 headers 0 lines) --- [May 13 12:31:34] NOTICE[2594]: chan_sip.c:24624 handle_response_register: Outbound Registration: Expiry for 157.xxx.x.xxx is 60 sec (Scheduling reregistration in 45 s) [May 13 12:31:34] Really destroying SIP dialog '6a07608e773eb2223f5417d4624abd15@192.xxx.x.xxx' Method: REGISTER [May 13 12:31:38] == Manager 'sendcron' logged on from 127.0.0.1 [May 13 12:31:38] -- Called 916015091286@default [May 13 12:31:38] -- Executing [916015091286@default:1] AGI("Local/916015091286@default-0000001e;2", "agi://127.0.0.1:4577/call_log") in new stack [May 13 12:31:38] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=001)) [May 13 12:31:38] -- <Local/916015091286@default-0000001e;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [May 13 12:31:38] -- Executing [916015091286@default:2] Set("Local/916015091286@default-0000001e;2", "CALLERID(ALL)="" <8336043243>") in new stack [May 13 12:31:38] -- Executing [916015091286@default:3] Dial("Local/916015091286@default-0000001e;2", "SIP/16015091286@ahsan,60,Tto") in new stack [May 13 12:31:38] == Using SIP RTP CoS mark 5 [May 13 12:31:38] Audio is at 14088 [May 13 12:31:38] Adding codec ulaw to SDP [May 13 12:31:38] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:31:38] Reliably Transmitting (NAT) to 157.xxx.x.xxx:5060: [May 13 12:31:38] INVITE sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:31:38] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK269364a1;rport [May 13 12:31:38] Max-Forwards: 70 [May 13 12:31:38] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as533dc261 [May 13 12:31:38] To: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:31:38] Contact: <sip:XXXXX@192.xxx.x.xxx:5060> [May 13 12:31:38] Call-ID: 69eb3e66659240c7207bf55840b2d8bb@192.xxx.x.xxx:5060 [May 13 12:31:38] CSeq: 102 INVITE [May 13 12:31:38] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:31:38] Date: Wed, 13 May 2020 16:31:38 GMT [May 13 12:31:38] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 13 12:31:38] Supported: replaces, timer [May 13 12:31:38] Remote-Party-ID: "8336043243" <sip:8336043243@192.xxx.x.xxx>;party=calling;privacy=off;screen=no [May 13 12:31:38] Content-Type: application/sdp [May 13 12:31:38] Content-Length: 257 [May 13 12:31:38] [May 13 12:31:38] v=0 [May 13 12:31:38] o=root 756778983 756778983 IN IP4 192.xxx.x.xxx [May 13 12:31:38] s=Asterisk PBX 13.29.2-vici [May 13 12:31:38] c=IN IP4 192.xxx.x.xxx

Sorry for the long output logs. couldn't find any discrepancies here except the CallerID part.

It is a new Install. I tried installing it again but the same issue is still there.

astersik version from /etc/astguiclient.conf
VARasterisk_version = 13

Admin >> Servers = 13.29.2-vici

Posted: **Wed May 13, 2020 1:49 pm**

Sorry, no time to read. I was hoping you'd DIFF or just find and post what was different between the two call types.

Also: Was this output from "asterisk-R" or "screen -r asterisk"? Note that "screen -r asterisk" shows AGI perl script errors that are NOT visible anywhere else. You can also enable agi debugging (just like sip debuggon) for even more information.

Posted: **Wed May 13, 2020 5:22 pm**

ok those logs are huge and entirely too much, u just needed from right b4 the call to right after LOL, but i went thru them and i notice 2 things, manual calls are restricting RTP(the audio) to a high port range over 20000, do u have your ports open for UDP from 8000-30000? Second thing is the manual calls are not spawning a local channel into dahdi before trying to connect the call which is being done thru the auto dialer. Auto is restricting RTP to like 8288 i think it was. So it could be the ports blocking the manual if theyre not open or something with the dialplan as it refers to manual dial dependig how your using it with the top method or below with override..... I would think its the ports....but im not 100% sure, im sure bill will be able to tell u more once he has the time to go thru those huge logs, in fact ill copy and paste what he needs to see to make it easier for him and repost that for u

Code: Select all: <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:10] SIP/2.0 183 Session Progress [May 13 12:28:10] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK7b18a359;received=103.8.112.182;rport=5060 [May 13 12:28:10] From: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:10] [May 13 12:28:10] v=0 [May 13 12:28:10] o=- 8216 8216 IN IP4 157.xxx.x.xxx [May 13 12:28:10] s=VOS3000 [May 13 12:28:10] c=IN IP4 157.xxx.x.xxx [May 13 12:28:10] t=0 0 [May 13 12:28:10] m=audio 23726 RTP/AVP 0 101 [May 13 12:28:10] a=rtpmap:0 PCMU/8000 [May 13 12:28:10] a=rtpmap:101 telephone-event/8000 [May 13 12:28:10] a=fmtp:101 0-15 [May 13 12:28:10] a=sendrecv [May 13 12:28:10] <-------------> [May 13 12:28:10] --- (11 headers 10 lines) --- [May 13 12:28:10] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:10] Found RTP audio format 0 [May 13 12:28:10] Found RTP audio format 101 [May 13 12:28:10] Found audio description format PCMU for ID 0 [May 13 12:28:10] Found audio description format telephone-event for ID 101 [May 13 12:28:10] Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [May 13 12:28:10] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 13 12:28:10] > 0x7f5d4400c860 -- Strict RTP learning after remote address set to: 157.xxx.x.xxx:23726 [May 13 12:28:10] Peer audio RTP is at port 157.xxx.x.xxx:23726 [May 13 12:28:10] -- SIP/ahsan-00000113 is making progress passing it to SIP/781-00000112 [May 13 12:28:10] Audio is at 17712 [May 13 12:28:10] Adding codec ulaw to SDP [May 13 12:28:10] Adding codec gsm to SDP [May 13 12:28:10] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:28:10] [May 13 12:28:10] <--- Transmitting (NAT) to xxx.xxx.x.xxx:10219 ---> [May 13 12:28:10] SIP/2.0 183 Session Progress [May 13 12:28:10] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-2c465720b60c3448-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:10] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:10] To: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:10] [May 13 12:28:10] v=0 [May 13 12:28:10] o=root 76119618 76119618 IN IP4 192.xxx.x.xxx [May 13 12:28:10] s=Asterisk PBX 13.29.2-vici [May 13 12:28:10] c=IN IP4 192.xxx.x.xxx [May 13 12:28:10] t=0 0 [May 13 12:28:10] m=audio 17712 RTP/AVP 0 3 101 [May 13 12:28:10] a=rtpmap:0 PCMU/8000 [May 13 12:28:10] a=rtpmap:3 GSM/8000 [May 13 12:28:10] a=rtpmap:101 telephone-event/8000 [May 13 12:28:10] a=fmtp:101 0-16 [May 13 12:28:10] a=ptime:20 [May 13 12:28:10] a=maxptime:150 [May 13 12:28:10] a=sendrecv [May 13 12:28:10] [May 13 12:28:10] <------------> [May 13 12:28:10] > 0x7f5d9002c5f0 -- Strict RTP switching to RTP target address xxx.xxx.x.xxx:8288 as source [May 13 12:28:10] > 0x7f5d4400c860 -- Strict RTP switching to RTP target address 157.xxx.x.xxx:23726 as source [May 13 12:28:15] > 0x7f5d4400c860 -- Strict RTP learning complete - Locking on source address 157.xxx.x.xxx:23726 [May 13 12:28:15] [May 13 12:28:15] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:23] v=0 [May 13 12:28:23] o=- 8216 8216 IN IP4 157.xxx.x.xxx [May 13 12:28:23] s=VOS3000 [May 13 12:28:23] c=IN IP4 157.xxx.x.xxx [May 13 12:28:23] t=0 0 [May 13 12:28:23] m=audio 23726 RTP/AVP 0 101 [May 13 12:28:23] a=rtpmap:0 PCMU/8000 [May 13 12:28:23] a=rtpmap:101 telephone-event/8000 [May 13 12:28:23] a=fmtp:101 0-15 [May 13 12:28:23] a=sendrecv [May 13 12:28:23] <-------------> [May 13 12:28:23] --- (14 headers 10 lines) --- [May 13 12:28:23] sip_route_dump: route/path hop: <sip:16015091286@157.xxx.x.xxx:5060> [May 13 12:28:23] Transmitting (NAT) to 157.xxx.x.xxx:5060: [May 13 12:28:23] ACK sip:16015091286@157.xxx.x.xxx:5060 SIP/2.0 [May 13 12:28:23] --- [May 13 12:28:23] -- SIP/ahsan-00000113 answered SIP/781-00000112 [May 13 12:28:23] Audio is at 17712 [May 13 12:28:23] Adding codec ulaw to SDP [May 13 12:28:23] Adding codec gsm to SDP [May 13 12:28:23] Adding non-codec 0x1 (telephone-event) to SDP [May 13 12:28:23] [May 13 12:28:23] <--- Reliably Transmitting (NAT) to xxx.xxx.x.xxx:10219 ---> [May 13 12:28:23] SIP/2.0 200 OK [May 13 12:28:23] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-2c465720b60c3448-1--d87543-;received=xxx.xxx.x.xxx;rport=10219 [May 13 12:28:23] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:23] To: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:23] [May 13 12:28:23] v=0 [May 13 12:28:23] o=root 76119618 76119618 IN IP4 192.xxx.x.xxx [May 13 12:28:23] s=Asterisk PBX 13.29.2-vici [May 13 12:28:23] c=IN IP4 192.xxx.x.xxx [May 13 12:28:23] t=0 0 [May 13 12:28:23] m=audio 17712 RTP/AVP 0 3 101 [May 13 12:28:23] a=rtpmap:0 PCMU/8000 [May 13 12:28:23] a=rtpmap:3 GSM/8000 [May 13 12:28:23] a=rtpmap:101 telephone-event/8000 [May 13 12:28:23] a=fmtp:101 0-16 [May 13 12:28:23] a=ptime:20 [May 13 12:28:23] a=maxptime:150 [May 13 12:28:23] a=sendrecv [May 13 12:28:23] [May 13 12:28:23] <------------> [May 13 12:28:23] -- Channel SIP/ahsan-00000113 joined 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:23] -- Channel SIP/781-00000112 joined 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:23] > 0x7f5d9002c5f0 -- Strict RTP learning complete - Locking on source address xxx.xxx.x.xxx:8288 [May 13 12:28:23] [May 13 12:28:23] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:23] ACK sip:916015091286@192.xxx.x.xxx:5060 SIP/2.0 [May 13 12:28:23] Via: SIP/2.0/UDP xxx.xxx.x.xxx:10219;branch=z9hG4bK-d87543-f66d0e26484a8263-1--d87543-;rport [May 13 12:28:23] Max-Forwards: 70 [May 13 12:28:23] Contact: <sip:781@xxx.xxx.x.xxx:10219> [May 13 12:28:23] To: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:23] From: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:23] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:23] CSeq: 2 ACK [May 13 12:28:23] User-Agent: eyeBeam release 3010n stamp 19039 [May 13 12:28:23] Content-Length: 0 [May 13 12:28:23] [May 13 12:28:23] <-------------> [May 13 12:28:23] --- (10 headers 0 lines) --- [May 13 12:28:24] [May 13 12:28:24] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:24] [May 13 12:28:24] <-------------> [May 13 12:28:31] Reliably Transmitting (NAT) to xxx.xxx.x.xxx:10219: [May 13 12:28:31] OPTIONS sip:781@xxx.xxx.x.xxx:10219 SIP/2.0 [May 13 12:28:31] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK2eb40e83;rport [May 13 12:28:31] Max-Forwards: 70 [May 13 12:28:31] From: "asterisk" <sip:asterisk@192.xxx.x.xxx>;tag=as5d0d4c20 [May 13 12:28:31] To: <sip:781@xxx.xxx.x.xxx:10219> [May 13 12:28:31] Contact: <sip:asterisk@192.xxx.x.xxx:5060> [May 13 12:28:31] [May 13 12:28:31] <--- SIP read from UDP:xxx.xxx.x.xxx:10219 ---> [May 13 12:28:31] SIP/2.0 200 OK [May 13 12:28:31] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK2eb40e83;rport=5060 [May 13 12:28:31] Contact: <sip:xxx.xxx.x.xxx:10219> [May 13 12:28:31] To: <sip:781@xxx.xxx.x.xxx:10219>;tag=6b76b048 [May 13 12:28:31] From: "asterisk"<sip:asterisk@192.xxx.x.xxx>;tag=as5d0d4c20 [May 13 12:28:31] Really destroying SIP dialog '17edc0194c11fe48254734c95200541c@192.xxx.x.xxx:5060' Method: OPTIONS [May 13 12:28:34] SIP/2.0 200 OK [May 13 12:28:34] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK318ce6b8;received=103.8.112.182;rport=5060 [May 13 12:28:34] From: <sip:XXXXX@157.xxx.x.xxx>;tag=as070a8fa3 [May 13 12:28:34] To: <sip:XXXXX@157.xxx.x.xxx> [May 13 12:28:34] NOTICE[2594]: chan_sip.c:24624 handle_response_register: Outbound Registration: Expiry for 157.xxx.x.xxx is 60 sec (Scheduling reregistration in 45 s) [May 13 12:28:34] Really destroying SIP dialog '6a07608e773eb2223f5417d4624abd15@192.xxx.x.xxx' Method: REGISTER [May 13 12:28:35] [May 13 12:28:35] <--- SIP read from UDP:157.xxx.x.xxx:5060 ---> [May 13 12:28:35] BYE sip:XXXXX@192.xxx.x.xxx:5060 SIP/2.0 [May 13 12:28:35] Via: SIP/2.0/UDP 157.xxx.x.xxx:5060;branch=z9hG4bK040135a26142b78e [May 13 12:28:35] Scheduling destruction of SIP dialog '3e1be6976fd0ddd76419ca9804201d46@192.xxx.x.xxx:5060' in 6400 ms (Method: BYE) [May 13 12:28:35] [May 13 12:28:35] <--- Transmitting (NAT) to 157.xxx.x.xxx:5060 ---> [May 13 12:28:35] SIP/2.0 200 OK [May 13 12:28:35] Via: SIP/2.0/UDP 157.xxx.x.xxx:5060;branch=z9hG4bK040135a26142b78e;received=157.xxx.x.xxx;rport=5060 [May 13 12:28:35] From: <sip:16015091286@157.xxx.x.xxx:5060>;tag=2808573f033d4d4d [May 13 12:28:35] To: <sip:XXXXX@192.xxx.x.xxx>;tag=as6b87b62d [May 13 12:28:35] -- Channel SIP/ahsan-00000113 left 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:35] -- Channel SIP/781-00000112 left 'simple_bridge' basic-bridge <a7508c5b-2d58-4698-9b59-9d8ec83c30ed> [May 13 12:28:35] == Spawn extension (default, 916015091286, 3) exited non-zero on 'SIP/781-00000112' [May 13 12:28:35] -- Executing [h@default:1] AGI("SIP/781-00000112", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----27-----11-----SIP 200 OK)") in new stack [May 13 12:28:35] -- <SIP/781-00000112>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----27-----11-----SIP 200 OK) completed, returning 0 [May 13 12:28:35] Scheduling destruction of SIP dialog '762d8d7406168d4d@SkQtUEM.' in 13376 ms (Method: ACK) [May 13 12:28:35] Reliably Transmitting (NAT) to xxx.xxx.x.xxx:10219: [May 13 12:28:35] BYE sip:781@xxx.xxx.x.xxx:10219 SIP/2.0 [May 13 12:28:35] Via: SIP/2.0/UDP 192.xxx.x.xxx:5060;branch=z9hG4bK65101929;rport [May 13 12:28:35] Max-Forwards: 70 [May 13 12:28:35] From: <sip:916015091286@192.xxx.x.xxx>;tag=as14ce0f8a [May 13 12:28:35] To: <sip:781@192.xxx.x.xxx>;tag=2d503a3f [May 13 12:28:35] Call-ID: 762d8d7406168d4d@SkQtUEM. [May 13 12:28:35] CSeq: 102 BYE [May 13 12:28:35] User-Agent: Asterisk PBX 13.29.2-vici [May 13 12:28:35] Proxy-Authorization: Digest username="781", realm="asterisk", algorithm=MD5, uri="sip:192.xxx.x.xxx", nonce="6d1aa3dc", response="a491dc8e8c77f25a22f946bef0a52be6" [May 13 12:28:35] X-Asterisk-HangupCause: Normal Clearing [May 13 12:28:35] X-Asterisk-HangupCauseCode: 16 [May 13 12:28:35] Content-Length: 0

^---thats the manual call kinda cleaned up a little bit...

Posted: **Wed May 13, 2020 5:40 pm**

You ignored this ...

williamconley wrote:... Was this output from "asterisk-R" or "screen -r asterisk"? Note that "screen -r asterisk" shows AGI perl script errors that are NOT visible anywhere else. You can also enable agi debugging (just like sip debuggon) for even more information.

It's not likely going to be the ports. To test, you could drop the firewall.

It could be related to the "local" call. Does that happen three times? If so, it could be the firewall blocking audio from an audio server from a different IP than the signalling server (port 5060 is for signalling, but the audio port could be anywhere from 10k-25k and may be from a different IP which could be blocked).

the question usually is how do these calls differ? Either in generation or at the moment of hangup. Someone requests the hangup ... that's the moment you want to compare between the servers. Right before the hangup, something happens to cause it, but that same thing doesn't happen on the successful call.

In a manual call, there is no script involved in this process as the call is generated pre-linked to the conference. But in an autodialed call the call must be connected to the conference after the answer occurs. That's where the "screen -r asterisk" comes in, tracking any errors from that script. It's also where agi debugging may come in handy as the script is an agi script. But that does not rule out a SIP-based reason which could be based on the details of the call generation. So attack from all angles.

Posted: **Wed May 13, 2020 7:29 pm**

Thanks for the help dear Much appreciated such a great community. @ carpenox & williamconley

The Output was from asterisk -r. I did try troubleshooting from the screen -r asterisk there was also no error there.
I also tried with no firewall the issue was still there.
I did try a new install without connecting to the internet before vicibox installation. This time it worked fine. now there is no issue in the autodialer as well.

Thank you

Posted: **Wed May 13, 2020 8:02 pm**

good to hear and thx for posting back ur good

vicidial.org

New Install, calls hangup

New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup

Re: New Install, calls hangup