dial internal extensions thru softphone

[jace]

Hi Guys,

we are trying to call internally using extensions thru softphones unfortunately there is no audio after the call has been established. But after putting the call on hold and resume it again, audio comes up and we're good to talk.

Is there a way to make this working without putting the calls on hold first for the audio to work? Please help.

Sorry, not sure if this is a vicidial issue. but thanks for the reply.

Thanks..

[williamconley]

That's an odd situation. Never even almost had that. LOL

Are the agents local to the vicidial server? (Same local subnet?) Are they accessing via the local subnet IP or the Public subnet IP?

Perhaps showing some asterisk CLI (and perhaps asterisk debug CLI) from a single call where this happens would be good ... (Please do this in a controlled situation, not 3000 lines of unrelated code from other calls ...).

[jace]

Hi William,

We do access our server remotely using the public IP.

We're getting these on asterisk cli:

[Nov 6 11:48:47] -- Executing [0011@default:1] Dial("SIP/1111-000057ab", "SIP/112|60|") in new stack
[Nov 6 11:48:47] -- Called 112
[Nov 6 11:48:48] -- SIP/112-000057ac is ringing
[Nov 6 11:48:55] -- SIP/112-000057ac answered SIP/1111-000057ab
[Nov 6 11:48:55] -- Packet2Packet bridging SIP/1111-000057ab and SIP/112-000057ac
[Nov 6 11:49:02] == Parsing '/etc/asterisk/manager.conf': [Nov 6 11:49:02] Found
[Nov 6 11:49:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 6 11:49:02] == Parsing '/etc/asterisk/manager.conf': [Nov 6 11:49:02] Found
[Nov 6 11:49:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 6 11:49:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 6 11:49:02] -- Started music on hold, class 'default', on SIP/112-000057ac
[Nov 6 11:49:03] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 6 11:49:04] -- Started music on hold, class 'default', on SIP/1111-000057ab
[Nov 6 11:49:04] -- Stopped music on hold on SIP/112-000057ac
[Nov 6 11:49:07] == Parsing '/etc/asterisk/manager.conf': [Nov 6 11:49:07] Found
[Nov 6 11:49:07] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 6 11:49:07] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 6 11:49:10] -- Stopped music on hold on SIP/1111-000057ab
[Nov 6 11:49:21] -- Executing [h@default:1] DeadAGI("SIP/1111-000057ab", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----34-----26") in new stack
[Nov 6 11:49:21] WARNING[406]: res_agi.c:2230 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI
[Nov 6 11:49:21] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -34-----26 completed, returning 0
[Nov 6 11:49:21] == Spawn extension (default, 0011, 1) exited non-zero on 'SIP/1111-000057ab'

Thanks,

[williamconley]

I'm a little confused on a couple points here.

vicibox redux 4.0.3; ...; Asterisk 1.4.44 from scratch.

1) If you installed with Vicibox ... why did you manually install asterisk? Or did you leave "from scratch" in there not knowing it meant "manually"?

Executing [0011@default:1] Dial("SIP/1111-000057ab", "SIP/112|60|") in new stack

2) It appears you dialed 0011 and then your sip account (1111) dialed sip account 112. But you did not dial 112. So ... where did it get 112 from? Is Phone Extension 112's dialplan number 0011?

Also, have you tried just waiting to see if sound will come on after the same amount of time whether you put them on hold or not?

[jace]

Hi William,

I'm very sorry for the confusion.

Yes, we'd used vicibox redux during the installation. Please disregard "from scratch" as I have just copied one of the signature here and was not able to delete it.

It appears you dialed 0011 and then your sip account (1111) dialed sip account 112. But you did not dial 112. So ... where did it get 112 from? Is Phone Extension 112's dialplan number 0011?

yes, 0011 is the dialplan of extension 112.

We have tried waiting several times as well but with no difference, audio still doesn't work unless we put it on hold and resume.

Thanks,