vicidial.org

When agents login, the softphone rings, connect to conference but no calls come through. I see that the leads are getting dialed but with no result. I checked the asterisk cli and see this...

channel.c:3612 __ast_request_and_dial: Unable to call channel Local/91XXXXXXXXXX@default

where XXXXXXXXXX is the phone number. I have setup a carrier from goautodial sip service. I hope the settings are correct. Any suggestions?

post your details for positive replies

1.Post the dialplan entry you are using to dial out
2. in cli typ sip show peers and sip show registry and make sure the carrier is registered properly

3. post a full cli of a single call.

1. Where will I find the dial plan entry? What CONTEXT should I be using when registering the SIP carrier.

See the log below...

Code: Select all

KY-UTP01*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
gs105/gs105                (Unspecified)    D   N      0        UNKNOWN
gs104/gs104                (Unspecified)    D   N      0        UNKNOWN
gs102/gs102                (Unspecified)    D   N      0        UNKNOWN
201/201                    10.1.0.100       D   N      10884    OK (268 ms)
goautodial/********      96.31.86.214         N      5060     OK (160 ms)
5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0 offline]
[Feb  9 15:28:56]   == Refreshing DNS lookups.
[Feb  9 15:29:02]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:29:02] Found
[Feb  9 15:29:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:29:02]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:29:02] Found
[Feb  9 15:29:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:29:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:29:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:29:07]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:29:07] Found
[Feb  9 15:29:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:29:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:29:08]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:29:08] Found
[Feb  9 15:29:08]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:29:08] NOTICE[28270]: chan_local.c:599 local_call: No such extension/context 912034882461@default while calling Local channel
[Feb  9 15:29:08] NOTICE[28270]: channel.c:3612 __ast_request_and_dial: Unable to call channel Local/912034882461@default
[Feb  9 15:29:10]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:30:01]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:30:01] Found
[Feb  9 15:30:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:30:01]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:30:01] Found
[Feb  9 15:30:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:30:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:30:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:30:06]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:30:06] Found
[Feb  9 15:30:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:30:06]   == Manager 'sendcron' logged off from 127.0.0.1
KY-UTP01*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
sip.goautodial.com:5060         **********         105 Registered           Thu, 09 Feb 2012 15:30:17
KY-UTP01*CLI> sip set debug peer 201
SIP Debugging Enabled for IP: 10.1.0.100:10884
[Feb  9 15:31:33] Reliably Transmitting (NAT) to 10.1.0.100:10884:
OPTIONS sip:201@10.1.0.100:10884;rinstance=498fb8ea74e19f64;cpd=on SIP/2.0
Via: SIP/2.0/UDP 172.16.0.127:5060;branch=z9hG4bK1633b686;rport
From: "asterisk" <sip:asterisk@172.16.0.127>;tag=as6d86a5a8
To: <sip:201@10.1.0.100:10884;rinstance=498fb8ea74e19f64;cpd=on>
Contact: <sip:asterisk@172.16.0.127>
Call-ID: 08b8c8c25cfe12f043aa508709c911b4@172.16.0.127
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 09 Feb 2012 15:31:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Feb  9 15:31:33]
<--- SIP read from 10.1.0.100:10884 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.127:5060;branch=z9hG4bK1633b686;rport=5060
Contact: <sip:10.1.0.100:10884>
To: <sip:201@10.1.0.100:10884;rinstance=498fb8ea74e19f64;cpd=on>;tag=c97ab840
From: "asterisk"<sip:asterisk@172.16.0.127>;tag=as6d86a5a8
Call-ID: 08b8c8c25cfe12f043aa508709c911b4@172.16.0.127
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
[Feb  9 15:31:33] --- (12 headers 0 lines) ---
[Feb  9 15:31:33] Really destroying SIP dialog '08b8c8c25cfe12f043aa508709c911b4@172.16.0.127' Method: OPTIONS
[Feb  9 15:31:33]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:31:33] Found
[Feb  9 15:31:33]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:31:33] NOTICE[28511]: chan_local.c:599 local_call: No such extension/context 912036458744@default while calling Local channel
[Feb  9 15:31:33] NOTICE[28511]: channel.c:3612 __ast_request_and_dial: Unable to call channel Local/912036458744@default
[Feb  9 15:31:35]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:31:55]
<--- SIP read from 10.1.0.100:10884 --->



<------------->
[Feb  9 15:32:01]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:32:01] Found
[Feb  9 15:32:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:32:01]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:32:01] Found
[Feb  9 15:32:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:32:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:32:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:32:02] NOTICE[2689]: chan_sip.c:8178 sip_reregister:    -- Re-registration for  4050113763@sip.goautodial.com
[Feb  9 15:32:02] NOTICE[2689]: chan_sip.c:13779 handle_response_register: Outbound Registration: Expiry for sip.goautodial.com is 120 sec (Scheduling reregistration in 105 s)
[Feb  9 15:32:06]   == Parsing '/etc/asterisk/manager.conf': [Feb  9 15:32:06] Found
[Feb  9 15:32:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb  9 15:32:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb  9 15:32:25]
<--- SIP read from 10.1.0.100:10884 --->



<------------->
[Feb  9 15:32:33] Reliably Transmitting (NAT) to 10.1.0.100:10884:
OPTIONS sip:201@10.1.0.100:10884;rinstance=498fb8ea74e19f64;cpd=on SIP/2.0
Via: SIP/2.0/UDP 172.16.0.127:5060;branch=z9hG4bK39f944bd;rport
From: "asterisk" <sip:asterisk@172.16.0.127>;tag=as52968cc6
To: <sip:201@10.1.0.100:10884;rinstance=498fb8ea74e19f64;cpd=on>
Contact: <sip:asterisk@172.16.0.127>
Call-ID: 188a290033bd1d2d0de42a8f07497181@172.16.0.127
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 09 Feb 2012 15:32:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Feb  9 15:32:33]
<--- SIP read from 10.1.0.100:10884 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.127:5060;branch=z9hG4bK39f944bd;rport=5060
Contact: <sip:10.1.0.100:10884>
To: <sip:201@10.1.0.100:10884;rinstance=498fb8ea74e19f64;cpd=on>;tag=3d3fbf47
From: "asterisk"<sip:asterisk@172.16.0.127>;tag=as52968cc6
Call-ID: 188a290033bd1d2d0de42a8f07497181@172.16.0.127
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->


Hi

just gothrough this link

http://goautodial.org/projects/goautodi ... rted_Guide

Thanks Stirker for pointing me in the right direction... I followed the steps and now I get this...

Code: Select all

[Feb 10 09:21:47]     -- Executing [912034885265@default:1] AGI("SIP/gs102-00000004", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 10 09:21:47]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 10 09:21:47]     -- Executing [912034885265@default:2] Dial("SIP/gs102-00000004", "SIP/12034885265@goautodial||tTo") in new stack
[Feb 10 09:21:47]     -- Called 12034885265@goautodial
[Feb 10 09:21:47] WARNING[27396]: channel.c:3908 ast_channel_make_compatible: No path to translate from SIP/goautodial-00000005(256) to SIP/gs102-00000004(4)
[Feb 10 09:21:47]   == Spawn extension (default, 912034885265, 2) exited non-zero on 'SIP/gs102-00000004'
[Feb 10 09:21:47]     -- Executing [h@default:1] DeadAGI("SIP/gs102-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CONGESTION----------") in new stack
[Feb 10 09:21:47]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CONGESTION---------- completed, returning 0


Striker,

I was successful installing the codec, and now can make outbound calls on that server. I did a fresh install on another server, settings are identical, but the call has no audio. Call rings on parties end, but when they answer there is no audio in either direction. On agents screen the i see Waiting for ring...1,2,3,etc then LIVE CALL, then quickly to CALL HUNGUP. Any input would be greatly appreciated. Thanks.

Code: Select all

[May 15 19:10:29]
<--- SIP read from 172.16.11.250:28442 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.20.8:5060;branch=z9hG4bK7b9d1d86;rport=5060
Contact: <sip:172.16.11.250:28442>
To: <sip:201@172.16.11.250:28442;rinstance=ede0e2b619e3beba;cpd=on>;tag=63f666ab
From: "asterisk"<sip:asterisk@172.16.20.8>;tag=as12ac0ff5
Call-ID: 3be780531c39951056d24a7b1aaec166@172.16.20.8
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0


<------------->
[May 15 19:10:29] --- (13 headers 0 lines) ---
[May 15 19:10:29] Really destroying SIP dialog '3be780531c39951056d24a7b1aaec166@172.16.20.8' Method: OPTIONS
[May 15 19:10:29]   == Parsing '/etc/asterisk/manager.conf': [May 15 19:10:29] Found
[May 15 19:10:29]   == Manager 'sendcron' logged on from 127.0.0.1
[May 15 19:10:29]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-7db8,2", "8600051|F") in new stack
[May 15 19:10:29]        > Channel Local/8600051@default-7db8,1 was answered.
[May 15 19:10:29]     -- Executing [92034333262@default:1] AGI("Local/8600051@default-7db8,1", "agi://127.0.0.1:4577/call_log") in new stack
[May 15 19:10:29]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 15 19:10:29]     -- Executing [92034333262@default:2] Dial("Local/8600051@default-7db8,1", "SIP/2034333262@png_sip_1||tTo") in new stack
[May 15 19:10:29] Audio is at 192.168.1.6 port 19862
[May 15 19:10:29] Adding codec 0x100 (g729) to SDP
[May 15 19:10:29] Adding codec 0x2 (gsm) to SDP
[May 15 19:10:29] Adding codec 0x4 (ulaw) to SDP
[May 15 19:10:29] Adding non-codec 0x1 (telephone-event) to SDP
[May 15 19:10:29] Reliably Transmitting (NAT) to 66.234.186.77:5060:
INVITE sip:2034333262@66.234.186.77;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK68b172cb;rport
From: "M5151510290000000029" <sip:0000000000@192.168.1.6>;tag=as4734aa06
To: <sip:2034333262@66.234.186.77;cpd=on>
Contact: <sip:0000000000@192.168.1.6>
Call-ID: 0b0a88fa61fe34d77676df915e0da828@192.168.1.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M5151510290000000029" <sip:0000000000@192.168.1.6>;privacy=off;screen=no
Date: Tue, 15 May 2012 19:10:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 2996 2996 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 19862 RTP/AVP 18 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[May 15 19:10:29]     -- Called 2034333262@png_sip_1
[May 15 19:10:30]
<--- SIP read from 66.234.186.77:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.6:5060;received=173.190.127.92;branch=z9hG4bK68b172cb;rport=5060
To: <sip:2034333262@66.234.186.77;cpd=on>
From: "M5151510290000000029" <sip:0000000000@192.168.1.6>;tag=as4734aa06
Call-ID: 0b0a88fa61fe34d77676df915e0da828@192.168.1.6
CSeq: 102 INVITE
Content-Length: 0


<------------->
[May 15 19:10:30] --- (7 headers 0 lines) ---
[May 15 19:10:31]   == Manager 'sendcron' logged off from 127.0.0.1
[May 15 19:10:32]
<--- SIP read from 66.234.186.77:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.6:5060;received=X.X.X.X;branch=z9hG4bK68b172cb;rport=5060
Record-Route: <sip:sansay1488457347rdb861@66.234.186.77:5060;lr;transport=udp>
To: <sip:2034333262@66.234.186.77;cpd=on>;tag=sansay1488457347rdb861
From: "M5151510290000000029" <sip:0000000000@192.168.1.6>;tag=as4734aa06
Call-ID: 0b0a88fa61fe34d77676df915e0da828@192.168.1.6
CSeq: 102 INVITE
Contact: <sip:2034333262@66.234.186.77:5060>
P-Asserted-Identity: <sip:2034333262@66.234.186.77;cpd=on>
Content-Type: application/sdp
Content-Length: 238

v=0
o=Sansay-VSXi 188 1 IN IP4 66.234.186.77
s=Session Controller
c=IN IP4 199.173.96.82
t=0 0
m=audio 51490 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[May 15 19:10:32] --- (11 headers 11 lines) ---
[May 15 19:10:32] Found RTP audio format 18
[May 15 19:10:32] Found RTP audio format 101
[May 15 19:10:32] Found audio description format G729 for ID 18
[May 15 19:10:32] Found audio description format telephone-event for ID 101
[May 15 19:10:32] Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[May 15 19:10:32] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 15 19:10:32] Peer audio RTP is at port 199.173.96.82:51490
[May 15 19:10:32]     -- SIP/png_sip_1-00000019 is making progress passing it to Local/8600051@default-7db8,1
[May 15 19:10:40]
<--- SIP read from 66.234.186.77:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;received=X.X.X.X;branch=z9hG4bK68b172cb;rport=5060
Record-Route: <sip:sansay1488457347rdb861@66.234.186.77:5060;lr;transport=udp>
To: <sip:2034333262@66.234.186.77;cpd=on>;tag=sansay1488457347rdb861
From: "M5151510290000000029" <sip:0000000000@192.168.1.6>;tag=as4734aa06
Call-ID: 0b0a88fa61fe34d77676df915e0da828@192.168.1.6
CSeq: 102 INVITE
P-Asserted-Identity: <sip:2034333262@66.234.186.77;cpd=on>
Contact: <sip:2034333262@66.234.186.77:5060>
Content-Type: application/sdp
Content-Length: 238

v=0
o=Sansay-VSXi 188 1 IN IP4 66.234.186.77
s=Session Controller
c=IN IP4 199.173.96.82
t=0 0
m=audio 51490 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[May 15 19:10:40] --- (11 headers 11 lines) ---
[May 15 19:10:40] Found RTP audio format 18
[May 15 19:10:40] Found RTP audio format 101
[May 15 19:10:40] Found audio description format G729 for ID 18
[May 15 19:10:40] Found audio description format telephone-event for ID 101
[May 15 19:10:40] Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[May 15 19:10:40] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 15 19:10:40] Peer audio RTP is at port 199.173.96.82:51490
[May 15 19:10:40] list_route: hop: <sip:sansay1488457347rdb861@66.234.186.77:5060;lr;transport=udp>
[May 15 19:10:40] set_destination: Parsing <sip:sansay1488457347rdb861@66.234.186.77:5060;lr;transport=udp> for address/port to send to
[May 15 19:10:40] set_destination: set destination to 66.234.186.77, port 5060
[May 15 19:10:40] Transmitting (NAT) to 66.234.186.77:5060:
ACK sip:2034333262@66.234.186.77:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK19cf97a2;rport
Route: <sip:sansay1488457347rdb861@66.234.186.77:5060;lr;transport=udp>
From: "M5151510290000000029" <sip:0000000000@192.168.1.6>;tag=as4734aa06
To: <sip:2034333262@66.234.186.77;cpd=on>;tag=sansay1488457347rdb861
Contact: <sip:0000000000@192.168.1.6>
Call-ID: 0b0a88fa61fe34d77676df915e0da828@192.168.1.6
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M5151510290000000029" <sip:0000000000@192.168.1.6>;privacy=off;screen=no
Content-Length: 0


1. check whether your headphone is working properly
2. if you are using any router , then open port 10000 to 20000

I am able to call between phones, and hear audio in both directions. Still unable to hear external call. I checked the router and port 10000-20000 is open. Using the same router i am able to make and hear a call using the first server we built.

I had to add my sip provider ip in the externip in sip.conf. Hope this helps.