Re: [asterisk-users] Add Monitor application to call suppresses audio
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David Backeberg wrote:
>> 5) exten => s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
>> 6) exten => s,n,Goto(s-${DIALSTATUS},1);
> What is the 6 for?
> What is the goto supposed to do?
Hi David.
The '6' is in case I get a "CHANUNAVAIL" or other error back from the
Dial command. If the call is connected then I never get to '6'.
I have determined that the only calls I seem to be having trouble
monitoring are the ones sent to my answering service. If I terminate
the call to my cell phone, my home POTS line, a POTS line here in the
office or even to the inbound PRI at the office, things work fine. I
can even record calls to the answering service's published number. It's
just when I go to the number assigned to us that there is trouble and
I'm currently chasing down the owner of that service to see exactly what
I'm dropping into there.
Thanks!
Barry
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Re: [asterisk-users] Add Monitor application to call suppresses audio
On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline wrote:
> If I insert a Monitor() prior to dialing the outbound call, I get no
> audio in the recording and the caller hears no audio. Occasionally it
> works (perhaps 1 out of 5 times) but most of the time the caller can't
> hear the callee, and vice versa.
>
> The fully working code looks like this:
> 1) exten => s,n(place),Verbose(4,Dialing answering service);
> 2) exten => s,n,Playback(vrec_prompts/this-call-may-be-recorded);
> 3) exten => s,n,Set(GROUP()=ANSSVC);
> 4) exten =>
> s,n,Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
> 5) exten => s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
> 6) exten => s,n,Goto(s-${DIALSTATUS},1);
What is the 6 for?
What is the goto supposed to do?
This could certainly explain why the first call works and not the
subsequent calls.
Why don't you want to just hangup the call after 5 completes?
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Re: [asterisk-users] Add Monitor application to call suppresses audio
This is getting really interesting. I had a chance to do some testing last night. To recap, here is what I'm attempting to do: Caller --> INBOUND_PRI --> Asterisk --> OUTBOUND_PRI --> AnswerService The caller dials our number, * picks the call and offers some choices. If the caller needs to speak to a human, * dials the answering service and then bridges the calls. It works flawlessly in this scenario. If I do Monitor() or MixMonitor() on the channel before dialing the outbound call, 99.9% of the time I get no audio to the caller. Now for the interesting part: If I substitute my cell phone number (xxx-) for that of the answering service (1-877-...) everything, including Monitor(), works just fine. The same goes for calling my home number instead of the cell phone. I don't know what my cell phone and home phone terminates into when I dial those numbers but if I call my answering service I know that they have a T1/PRI. I have done a PRI DEBUG SPAN 4 (where 4 is the outbound PRI) and repeated the tests, capturing the output. Nothing in them jumps out at me as yet but I'll keep looking at them. Does this offer anything suggestive David? Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add Monitor application to call suppresses audio
On Mon, May 18, 2009 at 5:16 PM, Barry L. Kline wrote: >> So you say call 1 with recording made a file, and the call connected with >> voice. >> And call2 with recording made a file, but the customer didn't hear the voice? > > Yes. In this case I'm using an outside Asterisk server to dial back in > and act like the customer. The first time there was a longer delay in > making the connection, but I was able to eventually hear the audio. > > The second attempt gave no audio to the customer side. Ridiculous question: if you do core show channels before first call after first call / before second call after second call Are the channels getting cleaned up properly? That is, are the hangups getting detected properly? There seems to be something weird with the call legs not lining up properly. The first call working and the second call not working is very suspicious. Similarly, can you put in two calls simultaneously? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add Monitor application to call suppresses audio
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Sorry for the delayed response, I was out of the office. David Backeberg wrote: > But there's not the native bridging status on the calls with recording > enabled, where as the native bridging report fires on the > recording-less dialplan. A clue perhaps... > So you say call 1 with recording made a file, and the call connected with > voice. > And call2 with recording made a file, but the customer didn't hear the voice? Yes. In this case I'm using an outside Asterisk server to dial back in and act like the customer. The first time there was a longer delay in making the connection, but I was able to eventually hear the audio. The second attempt gave no audio to the customer side. > What happens if you use MixMonitor() instead? > Are you mixing these calls back together afterward? My recollection is > that monitor makes a call in two halves, one for sender, and for > receiver, and then you have to multiplex the halves back together > afterwards. Are you doing the multiplex step? MixMonitor() doesn't act any differently... same no audio condition. Monitor does indeed act as you say but if you add an ',m' option to the call it will do the merge the files back together. I'm going to make my cell phone the target and see what the callee (the answering service) is hearing. Thanks for your comments David. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKEdAtCFu3bIiwtTARAgnnAKCn1tQKTT8/orBRRhsZ/EjgQ/0U9gCeJOXg yvDYr2t/iSG40J+7H4XLOf0= =KlGM -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add Monitor application to call suppresses audio
On Wed, May 13, 2009 at 11:00 PM, Barry L. Kline wrote: > The first call is the dialplan I have been using which works perfectly. > > The second call, which worked, was the first attempt to use Monitor(), > after having restarted Asterisk. > > The third call is another attempt at getting a recording. It, and any > subsequent call, fails miserably. > > I'm open for any suggestions. Thanks very much David. > > Barry In all cases, your dial argument to limit the call to 15 seconds seems to be working, and the calls are definitely getting hungup at 15 seconds. But there's not the native bridging status on the calls with recording enabled, where as the native bridging report fires on the recording-less dialplan. So you say call 1 with recording made a file, and the call connected with voice. And call2 with recording made a file, but the customer didn't hear the voice? What happens if you use MixMonitor() instead? Are you mixing these calls back together afterward? My recollection is that monitor makes a call in two halves, one for sender, and for receiver, and then you have to multiplex the halves back together afterwards. Are you doing the multiplex step? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add Monitor application to call suppresses audio
David Backeberg wrote: > I don't know why recording is breaking your calls. My guess is > something is screwed up with your PRI configuration. Are you getting > alarms in your logs from dahdi? Not a peep, either with or without using the monitor command. I've been using this system for around four months during which time it has performed flawlessly, running through 20K calls. > You should try to reproduce the problem on demand by generalizing your > dialplan, change the number of the answer service to the number of > your cell phone, and run some calls through. Done. > I've been recording calls with 1.6.0 series using MixMonitor() and > haven't been having problems, making me think the recordings step is > coincidental. Crank up the verbosity, run some calls through and tell > us what's happening. To avoid wrapping, I've posted the results from my tests to this link: http://www.pastebin.ca/1422291 The first call is the dialplan I have been using which works perfectly. The second call, which worked, was the first attempt to use Monitor(), after having restarted Asterisk. The third call is another attempt at getting a recording. It, and any subsequent call, fails miserably. I'm open for any suggestions. Thanks very much David. Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add Monitor application to call suppresses audio
On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline wrote: > If I insert a Monitor() prior to dialing the outbound call, I get no > audio in the recording and the caller hears no audio. Occasionally it > works (perhaps 1 out of 5 times) but most of the time the caller can't > hear the callee, and vice versa. > I'm using Asterisk 1.6.0.9, LIBPRI 1.4.10, and DAHDI 2.1.0.4. > Can anyone shine any light on why this problem is occurring? I don't know why recording is breaking your calls. My guess is something is screwed up with your PRI configuration. Are you getting alarms in your logs from dahdi? You should try to reproduce the problem on demand by generalizing your dialplan, change the number of the answer service to the number of your cell phone, and run some calls through. I've been recording calls with 1.6.0 series using MixMonitor() and haven't been having problems, making me think the recordings step is coincidental. Crank up the verbosity, run some calls through and tell us what's happening. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Add Monitor application to call suppresses audio
I have an application where we receive calls on an inbound PRI. After
hours, our Asterisk box dials our answering service on an outbound PRI
and then bridges the caller to the answering service. The flow looks
like this:
(CALLER)INBOUND_PRI --> CONTEXT --> GOSUB(Incoming) -->
GOSUB(bridge-to-anssrv) --> DIAL(answering_service) -->
OUTBOUND_PRI(service)
This has been working fine for months without so much as a burp. What I
need to do is record these calls.
If I insert a Monitor() prior to dialing the outbound call, I get no
audio in the recording and the caller hears no audio. Occasionally it
works (perhaps 1 out of 5 times) but most of the time the caller can't
hear the callee, and vice versa.
The fully working code looks like this:
1) exten => s,n(place),Verbose(4,Dialing answering service);
2) exten => s,n,Playback(vrec_prompts/this-call-may-be-recorded);
3) exten => s,n,Set(GROUP()=ANSSVC);
4) exten =>
s,n,Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
5) exten => s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten => s,n,Goto(s-${DIALSTATUS},1);
If I insert
exten => s,n,Monitor(wav,${CALLFILENAME},m);
before the dial command on line 5, I'm virtually guaranteed that the
call will fail and no audio will be passed.
I'm using Asterisk 1.6.0.9, LIBPRI 1.4.10, and DAHDI 2.1.0.4.
Can anyone shine any light on why this problem is occurring?
TIA,
Barry
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