GoAutodial can't hear the voice prompt

[chinita17]

Hi guys another help pls....


Customer and agent can't hear each other on the line. What the possible cause of it?

Here's the CLI result:

[Apr 25 08:37:39] -- Executing [8005@default:1] Dial("SIP/8001-00000003", "SIP/8005|60|") in new stack
[Apr 25 08:37:39] WARNING[29532]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Apr 25 08:37:39] == Everyone is busy/congested at this time (1:0/0/1)
[Apr 25 08:37:39] -- Executing [8005@default:2] Goto("SIP/8001-00000003", "default|850266666666668005|1") in new stack
[Apr 25 08:37:39] -- Goto (default,850266666666668005,1)
[Apr 25 08:37:39] -- Executing [850266666666668005@default:1] Wait("SIP/8001-00000003", "1") in new stack
[Apr 25 08:37:40] -- Executing [850266666666668005@default:2] VoiceMail("SIP/8001-00000003", "8005|u") in new stack
[Apr 25 08:37:40] -- <SIP/8001-00000003> Playing 'vm-theperson' (language 'en')
[Apr 25 08:37:41] -- <SIP/8001-00000003> Playing 'digits/8' (language 'en')
[Apr 25 08:37:42] -- <SIP/8001-00000003> Playing 'digits/0' (language 'en')
[Apr 25 08:37:43] -- <SIP/8001-00000003> Playing 'digits/0' (language 'en')
[Apr 25 08:37:44] -- <SIP/8001-00000003> Playing 'digits/5' (language 'en')
[Apr 25 08:37:44] -- <SIP/8001-00000003> Playing 'vm-isunavail' (language 'en')
[Apr 25 08:37:45] -- <SIP/8001-00000003> Playing 'vm-intro' (language 'en')
[Apr 25 08:37:51] -- <SIP/8001-00000003> Playing 'beep' (language 'en')
[Apr 25 08:37:51] -- Recording the message
[Apr 25 08:37:51] -- x=0, open writing: /var/spool/asterisk/voicemail/default/8005/tmp/ED4qtD format: wav49, 0x8414300
[Apr 25 08:37:51] -- x=1, open writing: /var/spool/asterisk/voicemail/default/8005/tmp/ED4qtD format: gsm, 0x863d178
[Apr 25 08:37:51] -- x=2, open writing: /var/spool/asterisk/voicemail/default/8005/tmp/ED4qtD format: wav, 0x8413cb8
[Apr 25 08:37:55] WARNING[29532]: app.c:615 __ast_play_and_record: No audio available on SIP/8001-00000003??
[Apr 25 08:37:55] -- User hung up
[Apr 25 08:37:55] == Spawn extension (default, 850266666666668005, 2) exited non-zero on 'SIP/8001-00000003'
[Apr 25 08:37:55] -- Executing [h@default:1] DeadAGI("SIP/8001-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Apr 25 08:37:55] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

But i don't hear on my softphone saying that the 8005 extension is unavailable unlike before. Weird, i do hear it before but after 6 or 7 days no voice prompt came out.

[williamconley]

Unable to create channel of type 'SIP'

This is not a good sign. When did this begin happening? Did you modify your configuration?

What Exactly are you trying to do when this occurs (remember this is a technical support forum, so a technical answer including button by button actions is in order as opposed to a statement like "I was trying to make a call" which does not tell me precisely what is happening ...).

[chinita17]

yes i do modify my configuration before for my extension.conf because i do experience the meetme not connecting to the sip phone. after adding the meetme portion on extension.conf that is missing the connection been ok and can establish a call. but after 6 or 7 days upon checking to run for a beta test for production can't hear any voice prompt even the other side of the line.

no other changes happen after then...

i can do an outbound call . where i'm just concerned i can't hear anything from the other line after a ring, no voice prompt like the extension number is unavailable. i'm tried it doing the direct dial from my softphone.

here's the cli result for outside number but can't hear anything after the customer answered the phone:

[Apr 25 10:06:23] --- (7 headers 0 lines) ---
[Apr 25 10:06:23] Really destroying SIP dialog '2cf7a7d9601dc95e5134f86955909100@192.168.1.1' Method: OPTIONS
[Apr 25 10:07:01] == Parsing '/etc/asterisk/manager.conf': [Apr 25 10:07:01] Found
[Apr 25 10:07:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 25 10:07:01] == Parsing '/etc/asterisk/manager.conf': [Apr 25 10:07:01] Found
[Apr 25 10:07:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 25 10:07:02] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 25 10:07:04] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 25 10:07:07] == Parsing '/etc/asterisk/manager.conf': [Apr 25 10:07:07] Found
[Apr 25 10:07:07] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 25 10:07:07] -- Executing [17024256201@default:1] AGI("SIP/8001-00000002", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 25 10:07:07] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 25 10:07:07] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 25 10:07:07] -- Executing [17024256201@default:2] Dial("SIP/8001-00000002", "SIP/17024256201@eastwest||tToR") in new stack
[Apr 25 10:07:07] Audio is at 192.168.1.1 port 17476
[Apr 25 10:07:07] Adding codec 0x4 (ulaw) to SDP
[Apr 25 10:07:07] Adding codec 0x100 (g729) to SDP
[Apr 25 10:07:07] Adding codec 0x2 (gsm) to SDP
[Apr 25 10:07:07] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 25 10:07:07] Reliably Transmitting (NAT) to 69.26.183.13:5060:
INVITE sip:17024256201@69.26.183.13;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK7da9e0de;rport
From: "8001" <sip:0000000000@192.168.1.1>;tag=as63c04a5b
To: <sip:17024256201@69.26.183.13;cpd=on>
Contact: <sip:0000000000@192.168.1.1>
Call-ID: 24decc0f2a88744131907d28564b7255@192.168.1.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8001" <sip:0000000000@192.168.1.1>;privacy=off;screen=no
Date: Wed, 25 Apr 2012 14:07:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 3207 3207 IN IP4 192.168.1.1
s=session
c=IN IP4 192.168.1.1
t=0 0
m=audio 17476 RTP/AVP 0 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Apr 25 10:07:07] -- Called 17024256201@eastwest
[Apr 25 10:07:07]
<--- SIP read from 69.26.183.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1:5060;received=67.207.165.226;branch=z9hG4bK7da9e0de;rport=5060
To: <sip:17024256201@69.26.183.13;cpd=on>
From: "8001" <sip:0000000000@192.168.1.1>;tag=as63c04a5b
Call-ID: 24decc0f2a88744131907d28564b7255@192.168.1.1
CSeq: 102 INVITE
Content-Length: 0


<------------->
[Apr 25 10:07:07] --- (7 headers 0 lines) ---
[Apr 25 10:07:10]
<--- SIP read from 69.26.183.13:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.1:5060;received=67.207.165.226;branch=z9hG4bK7da9e0de;rport=5060
Record-Route: <sip:sansay639401363rdb5969@69.26.183.13:5060;lr;transport=udp>
To: <sip:17024256201@69.26.183.13;cpd=on>;tag=sansay639401363rdb5969
From: "8001" <sip:0000000000@192.168.1.1>;tag=as63c04a5b
Call-ID: 24decc0f2a88744131907d28564b7255@192.168.1.1
CSeq: 102 INVITE
Contact: <sip:17024256201@69.26.183.13:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Disposition: session; handling=required
Content-Type: application/sdp
Content-Length: 226

v=0
o=Sansay-VSXi 188 1 IN IP4 69.26.183.13
s=Session Controller
c=IN IP4 173.245.44.23
t=0 0
m=audio 12112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

<------------->
[Apr 25 10:07:10] --- (12 headers 11 lines) ---
[Apr 25 10:07:10] Found RTP audio format 0
[Apr 25 10:07:10] Found RTP audio format 101
[Apr 25 10:07:10] Found audio description format PCMU for ID 0
[Apr 25 10:07:10] Found audio description format telephone-event for ID 101
[Apr 25 10:07:10] Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Apr 25 10:07:10] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 25 10:07:10] Peer audio RTP is at port 173.245.44.23:12112
[Apr 25 10:07:10] -- SIP/eastwest-00000003 is making progress passing it to SIP/8001-00000002
[Apr 25 10:07:23] Reliably Transmitting (NAT) to 69.26.183.13:5060:
OPTIONS sip:69.26.183.13;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK26c8bffd;rport
From: "asterisk" <sip:asterisk@192.168.1.1>;tag=as1a92da84
To: <sip:69.26.183.13;cpd=on>
Contact: <sip:asterisk@192.168.1.1>
Call-ID: 685e79b7487c431619b13b7f6f41cc7f@192.168.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 25 Apr 2012 14:07:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Apr 25 10:07:23]
<--- SIP read from 69.26.183.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;received=67.207.165.226;branch=z9hG4bK26c8bffd;rport=5060
To: <sip:69.26.183.13;cpd=on>
From: "asterisk" <sip:asterisk@192.168.1.1>;tag=as1a92da84
Call-ID: 685e79b7487c431619b13b7f6f41cc7f@192.168.1.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
[Apr 25 10:07:23] --- (7 headers 0 lines) ---
[Apr 25 10:07:23] Really destroying SIP dialog '685e79b7487c431619b13b7f6f41cc7f@192.168.1.1' Method: OPTIONS
[Apr 25 10:07:37]
<--- SIP read from 69.26.183.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;received=67.207.165.226;branch=z9hG4bK7da9e0de;rport=5060
Record-Route: <sip:sansay639401363rdb5969@69.26.183.13:5060;lr;transport=udp>
To: <sip:17024256201@69.26.183.13;cpd=on>;tag=sansay639401363rdb5969
From: "8001" <sip:0000000000@192.168.1.1>;tag=as63c04a5b
Call-ID: 24decc0f2a88744131907d28564b7255@192.168.1.1
CSeq: 102 INVITE
Supported: 100rel
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Disposition: session; handling=required
Contact: <sip:17024256201@69.26.183.13:5060>
Content-Type: application/sdp
Content-Length: 226

v=0
o=Sansay-VSXi 188 1 IN IP4 69.26.183.13
s=Session Controller
c=IN IP4 173.245.44.23
t=0 0
m=audio 12112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

<------------->
[Apr 25 10:07:37] --- (14 headers 11 lines) ---
[Apr 25 10:07:37] Found RTP audio format 0
[Apr 25 10:07:37] Found RTP audio format 101
[Apr 25 10:07:37] Found audio description format PCMU for ID 0
[Apr 25 10:07:37] Found audio description format telephone-event for ID 101
[Apr 25 10:07:37] Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Apr 25 10:07:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 25 10:07:37] Peer audio RTP is at port 173.245.44.23:12112
[Apr 25 10:07:37] list_route: hop: <sip:sansay639401363rdb5969@69.26.183.13:5060;lr;transport=udp>
[Apr 25 10:07:37] set_destination: Parsing <sip:sansay639401363rdb5969@69.26.183.13:5060;lr;transport=udp> for address/port to send to
[Apr 25 10:07:37] set_destination: set destination to 69.26.183.13, port 5060
[Apr 25 10:07:37] Transmitting (NAT) to 69.26.183.13:5060:
ACK sip:17024256201@69.26.183.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK67d8f072;rport
Route: <sip:sansay639401363rdb5969@69.26.183.13:5060;lr;transport=udp>
From: "8001" <sip:0000000000@192.168.1.1>;tag=as63c04a5b
To: <sip:17024256201@69.26.183.13;cpd=on>;tag=sansay639401363rdb5969
Contact: <sip:0000000000@192.168.1.1>
Call-ID: 24decc0f2a88744131907d28564b7255@192.168.1.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8001" <sip:0000000000@192.168.1.1>;privacy=off;screen=no
Content-Length: 0


---
[Apr 25 10:07:37] -- SIP/eastwest-00000003 answered SIP/8001-00000002

[dxtrqnt108]

hi guys
i have similar issue here with goautodial that i installed from scratch using the same disk used in a different office i set up a few months back.
asterisk
VERSION: 2.4-309a
BUILD: 110430-1642
© 2011 ViciDial Group
goautodial 2.1
single NIC
AMD Phenom(tm) II X4 925 Processor

working with SIP provider.. the outbound is now doing great on both auto/manual dial.
i created 2 campaigns; the opener (prospecting) and the closers inbound/outbound mix (sales)... openers are doing outbound autodial with fresh leads and all the prospected leads will be moved to the sales campaign and the sales agents will process those leads in inbound_man dialing method. when sales manual dial the leads, its doing great but if somebody calls in, cant hear anything and call drops after 20+ secs. tried dialing our DIDmyself, cant hear on both ends.
here is whats happening in the CLI dialing the DID

[Oct 18 13:05:58] -- Executing [8888939336@trunkinbound:1] AGI("SIP/APN-0000 03b3", "agi-DID_route.agi") in new stack
[Oct 18 13:05:58] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_r oute.agi
[Oct 18 13:05:58] -- AGI Script Executing Application: (Monitor) Options: (w av|/var/spool/asterisk/monitor/MIX/20121018130558_8888939336_Anonymous)
[Oct 18 13:05:58] -- AGI Script agi-DID_route.agi completed, returning 0
[Oct 18 13:05:58] -- Executing [8018@default:1] Dial("SIP/APN-000003b3", "SI P/8018|60|") in new stack
[Oct 18 13:05:58] -- Called 8018
[Oct 18 13:05:58] -- SIP/8018-000003b4 is ringing
[Oct 18 13:05:58] -- SIP/8018-000003b4 is ringing
[Oct 18 13:06:02] == Parsing '/etc/asterisk/manager.conf': [Oct 18 13:06:02] F ound
[Oct 18 13:06:02] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 18 13:06:02] == Parsing '/etc/asterisk/manager.conf': [Oct 18 13:06:02] F ound
[Oct 18 13:06:02] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 18 13:06:02] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 18 13:06:03] -- SIP/8018-000003b4 answered SIP/APN-000003b3
[Oct 18 13:06:04] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 18 13:06:07] == Parsing '/etc/asterisk/manager.conf': [Oct 18 13:06:07] F ound

[williamconley]

goautodial is not a scratch installation. it is an automated .iso installation. scratch would be where you installed a linux OS and then installed all the packages to make vicidial work yourself at the command line.

please post the name of the .iso installer (the goautodial disk). while "the same disk i used before" may help you to know which one, it doesn't really help the rest of us that much. LOL

did you use the vicidial managers manual to set up your inbound route? if not, please do! that's what it is for.

you have done well getting it to push to the agi-DID_route, but after that it is recommended to use the manager's manual to practice routing it for use. I generally recommend routing it to an ingroup before anything else (this provides Music on Hold to demonstrate success, which is helpful). Then when you figure out how to log an agent into that ingroup ... your cycle is complete.