outgoing call

[JCSANHUEZA]

the call is received by the agent but ringing

outgoing call

Campaign ID: TESTCAMP
Campaign Name: Test campaign for VICIdial
Active: Y
Allow Closers: Y
Hopper Level: 5
Dial Method: RATIO
Auto Dial Level: 1
Next Agent Call: oldest call finish
Local Call Time: 9am-9pm

Thanks in Advance

[Crimson Jack]

I think people are going to need more information on your settings than what youve provided.

[DomeDan]

@JCSANHUEZA
Post things like dial-plan, account entry and output from the asterisk CLI when this happen.
look at the CLI in a controlled enviroment, only one logged in agent and reproduce the problem with one call and then post the output here.

you might also want to check our similar posts like:
"Agent hear ringing in auto dial mode" http://www.vicidial.org/VICIDIALforum/v ... hp?t=20213
"After changing from T1's to SIP - agents get ringing calls" http://www.vicidial.org/VICIDIALforum/v ... php?t=8954
"Ring For Every call In Outbound Auto Dialed Campaign" http://www.vicidial.org/VICIDIALforum/v ... hp?t=19653
"Only Ringing when On Dialer But Connecting when Manually" http://www.vicidial.org/VICIDIALforum/v ... hp?t=14644

[JCSANHUEZA]

my sip trunk

[siptrunk]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
type=peer
host=sip.trunk.server.com
dtmfmode=auto
context=trunkinbound

TRUNK = SIP/siptrunk

exten => _01XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _01XXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:0},,tTor)
exten => _01XXXXXXXXXX,3,Hangup

[mcargile]

Try removing the 't' and the 'r' option from the dial command. They are not needed, and can cause some issues at times.

Please post the output from the Asterisk CLI while one of these calls is going out.

This issue might be with the upstream SIP server. I had a client having this issue, and come to find out it was because the SIP devise they were dialing through had 96 analog phone lines connected to via channel banks rather than some sort of digital circuit. When the devise placed an outbound call, it would immediately send a 200 OK to the Vicidial server even though the analog line was ringing.

[striker]

hi

are you using sip(VOIP) trunk to make outgoing calls or you are using the sangoma cards for outgoing.
as i can see in your singnature sangoma A104d mentioned.

[JCSANHUEZA]

Try removing the 't' and the 'r' option from the dial command. They are not needed, and can cause some issues at times.

Please post the output from the Asterisk CLI while one of these calls is going out.

This issue might be with the upstream SIP server. I had a client having this issue, and come to find out it was because the SIP devise they were dialing through had 96 analog phone lines connected to via channel banks rather than some sort of digital circuit. When the devise placed an outbound call, it would immediately send a 200 OK to the Vicidial server even though the analog line was ringing.

This is CLI for one call .....in this case the call is delivered to the agent but still ringing

Code: Select all

[Dec 29 08:41:41]   == Manager 'sendcron' logged on from 127.0.0.1
[Dec 29 08:41:43]   == Manager 'sendcron' logged off from 127.0.0.1
[Dec 29 08:41:43]   == Manager 'sendcron' logged off from 127.0.0.1
[Dec 29 08:41:53]   == Parsing '/etc/asterisk/manager.conf': [Dec 29 08:41:53] Found
[Dec 29 08:41:53]   == Manager 'sendcron' logged on from 127.0.0.1
[Dec 29 08:41:53]     -- Executing [528714552357@default:1] AGI("Local/528714552357@default-f034,2", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 29 08:41:53]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 29 08:41:53]     -- Executing [528714552357@default:2] Dial("Local/528714552357@default-f034,2", "SIP/grandstream/528714552357||To") in new stack
[Dec 29 08:41:53]     -- Called grandstream/528714552357
[Dec 29 08:41:53]     -- SIP/grandstream-00000003 answered Local/528714552357@default-f034,2
[Dec 29 08:41:53]        > Channel Local/528714552357@default-f034,1 was answered.
[Dec 29 08:41:53]     -- Executing [8368@default:1] Playback("Local/528714552357@default-f034,1", "sip-silence") in new stack
[Dec 29 08:41:53]     -- <Local/528714552357@default-f034,1> Playing 'sip-silence' (language 'en')
[Dec 29 08:41:53] WARNING[6534]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Dec 29 08:41:53]     -- Executing [8368@default:2] AGI("Local/528714552357@default-f034,1", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 29 08:41:53]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 29 08:41:53]     -- Executing [8368@default:3] AGI("Local/528714552357@default-f034,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Dec 29 08:41:53]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 29 08:41:53]     -- Executing [h@default:1] DeadAGI("Local/528714552357@default-f034,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0") in new stack
[Dec 29 08:41:54]     -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Dec 29 08:41:54]     -- Executing [8368@default:4] AGI("SIP/grandstream-00000003", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Dec 29 08:41:54]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 29 08:41:54]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0 completed, returning 0
[Dec 29 08:41:54]   == Spawn extension (default, 528714552357, 2) exited non-zero on 'Local/528714552357@default-f034,2'
[Dec 29 08:41:54] ERROR[6534]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Dec 29 08:41:54]     -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Dec 29 08:41:54]     -- Executing [192*168*001*200*8600051@default:1] Goto("SIP/grandstream-00000003", "default|8600051|1") in new stack
[Dec 29 08:41:54]     -- Goto (default,8600051,1)
[Dec 29 08:41:54]     -- Executing [8600051@default:1] MeetMe("SIP/grandstream-00000003", "8600051|F") in new stack
[Dec 29 08:41:55]   == Manager 'sendcron' logged off from 127.0.0.1

Other Call..... in this case the call is answered but not sent to the agent.

Code: Select all

[Dec 29 08:52:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Dec 29 08:52:02]     -- SIP/nextor-0000000b answered Local/528717051550@default-1af5,2
[Dec 29 08:52:02]        > Channel Local/528717051550@default-1af5,1 was answered.
[Dec 29 08:52:02]     -- Executing [8368@default:1] Playback("Local/528717051550@default-1af5,1", "sip-silence") in new stack
[Dec 29 08:52:02]     -- <Local/528717051550@default-1af5,1> Playing 'sip-silence' (language 'en')
[Dec 29 08:52:02] WARNING[7590]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Dec 29 08:52:02]     -- Executing [8368@default:2] AGI("Local/528717051550@default-1af5,1", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 29 08:52:02]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 29 08:52:02]     -- Executing [8368@default:3] AGI("Local/528717051550@default-1af5,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Dec 29 08:52:02]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 29 08:52:02]     -- Executing [h@default:1] DeadAGI("Local/528717051550@default-1af5,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0") in new stack
[Dec 29 08:52:02] WARNING[7563]: res_agi.c:2212 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI
[Dec 29 08:52:03]     -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Dec 29 08:52:03]     -- Executing [8368@default:4] AGI("Local/528717051550@default-1af5,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Dec 29 08:52:03]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 29 08:52:03]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0 completed, returning 0
[Dec 29 08:52:03]   == Spawn extension (default, 528717051550, 2) exited non-zero on 'Local/528717051550@default-1af5,2'
[Dec 29 08:52:03]   == Spawn extension (default, 8368, 4) exited non-zero on 'Local/528717051550@default-1af5,1'
[Dec 29 08:52:03]     -- Executing [h@default:1] DeadAGI("Local/528717051550@default-1af5,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Dec 29 08:52:04]   == Manager 'sendcron' logged off from 127.0.0.1
[Dec 29 08:52:04]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Dec 29 08:52:07]   == Parsing '/etc/asterisk/manager.conf': [Dec 29 08:52:07] Found

[JCSANHUEZA]

hi

are you using sip(VOIP) trunk to make outgoing calls or you are using the sangoma cards for outgoing.
as i can see in your singnature sangoma A104d mentioned.

I am using sip trunk to make outgoing call.... the samgoma hardware have been removed.

Thanks in advance.

[DomeDan]

print the output from
dialplan show 528714552357@default
(run it in the asterisk CLI)

[williamconley]

If you are using a grandstream gateway, you must have that configured properly before you can begin. Try dialing through it directly with a soft phone, and when you get that working you can modify the system to mimic your soft phone method.

Any trunk in asterisk will work in Vicidial, but that requires the trunk be functional in asterisk before you begin. If you work through support with grandstream it is usually not particularly difficult to get working, but it is not "simple" either. I do like the grandstream gateways, if you are using one, I highly recommend them as an excellent alternative to a 4 or 8 port voip card.