* You are viewing the archive for May 3rd, 2011

How to debug MixMonitor misbehaviour

Thanks for the input.

Yes, I did call out many times, but the recording doesn’t happen even after
the call is bridged and there is two way audio. I also took out the “b”
option and so it should recording the ringing right (even before call is
bridged) but it doesn’t do that or any recording at all.

Any other suggestions as to what I can do to see why this is not recording?

Regards,

On Tue, May 3, 2011 at 2:13 AM, virendra bhati wrote:

> Hi,
>
> As per your Dialplan MixMonitor will work after call bridge, In you case
> still call is not bridge. That’s why MixMonitor is waiting of call bridge…
>
> *
> MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)
> option b=>** A bridge flag allows recording to only take place when the
> channel is bridged.*
>
> So just for test make sip call and start mixmonitor to test the recorded
> file.
> default path od recording id
> *
> /var/spool/asterisk/monitor/
>
> *
> On Tue, May 3, 2011 at 10:40 AM, Bruce B wrote:
>
>> Hi everyone,
>>
>> For some reason MixMonitor doesn’t record when it should; It actually
>> shows the MixMonitor line just fine on the CLI. How can MixMonitor be
>> debugged for things like privilege issues or filename issues?
>>
>> **I had this working at one point and then stopped working. Not sure what
>> I changed.
>>
>> System Info:
>> Asterisk 1.4.21.2
>> Queuemetrics 1.6.3.0
>>
>>
>> [queuedial]
>> ; this piece of dialplan is just a calling hook into the [qm-queuedial]
>> context that actually does the
>> ; outbound dialing – replace as needed – just fill in the same variables.
>> exten => _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3})
>> exten => _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3})
>> exten => _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)})
>> exten => _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER})
>> exten => _XXX.,n,Set(QueueName=${QDIALER_QUEUE})
>> *exten => _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)*
>> exten => _XXX.,n,Goto(qm-queuedial,s,1)
>>
>> CLI output:
>> — Called 4904166356574@queuedial/n
>> — Executing [4904166356574@queuedial:1]
>> Set(“Local/4904166356574@queuedial-d851,2″, “QDIALER_QUEUE=q-490″) in new
>> stack
>> — Executing [4904166356574@queuedial:2]
>> Set(“Local/4904166356574@queuedial-d851,2″, “QDIALER_NUMBER=4166356574″)
>> in new stack
>> — Executing [4904166356574@queuedial:3]
>> Set(“Local/4904166356574@queuedial-d851,2″,
>> “QDIALER_AGENT=Agent/19053640558″) in new stack
>> — Executing [4904166356574@queuedial:4]
>> Set(“Local/4904166356574@queuedial-d851,2″,
>> “QDIALER_CHANNEL=ZAP/g0/4166356574″) in new stack
>> — Executing [4904166356574@queuedial:5]
>> Set(“Local/4904166356574@queuedial-d851,2″, “QueueName=q-490″) in new
>> stack
>> * — Executing [4904166356574@queuedial:6]
>> MixMonitor(“Local/4904166356574@queuedial-d851,2″,
>> “Q-q-490-1304399098.18.WAV|b|”) in new stack*
>> — Executing [4904166356574@queuedial:7]
>> Goto(“Local/4904166356574@queuedial-d851,2″, “qm-queuedial|s|1″) in new
>> stack
>> — Goto (qm-queuedial,s,1)
>>
>> Trying to locate file:
>> root@pbx:~ $ updatedb
>> root@pbx:~ $ locate Q-q-490-1304399098.18.WAV
>> root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q*
>> ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory
>>
>> I also turned on the Debug but I couldn’t see anything out of the norm. As
>> you can see above the CLI output is just fine.
>>
>> Thanks,
>> Bruce
>>
>> –
>> _____________________________________________________________________
>> — Bandwidth and Colocation Provided by http://www.api-digital.com
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> –
>
>
>
> —–
> Thanks and regards
>
> Virendra Bhati
> +91-9172341457
>
>
> –
> _____________________________________________________________________
> — Bandwidth and Colocation Provided by http://www.api-digital.com
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

asterisk call forwarding

Thank you so much that solved my database issue. Now how asterisk will
forward call ?

Or I need to specify gotoif statment in my stdexten to check database
key and take action?

Having redundancy, so if first IP failed then send for the other

Hi All;

I need to configure the SIP account so if first IP address failed then to send for the second IP address. How to do this?

While configuring the sip account, at the host parameter, can I give two IP addresses separated by comma? Or what should I do to have such redundancy?

Regards
Bilal

receive faxes

does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i’ve tried using “app_fax.so” with T38 but i keep
getting “Transmission failed”

dial from voicemail

Hi
Is it possible to dial from within voicemail to reach another extension.
I would like my customers to have a choice of dialing 1 to get my cell
phone while in voicemail or to just leave a message at the tone.

Thanks

Kelly

Asterisk 1.6 Questions

I have a couple of questions about asterisk 1.6:

Can you change codecs mid-call upon re-invite?

Can you handle the SDP offer-answer in the 200-ACK instead of the usual
INVITE-200?

Thanks in advance for any insight.

Gary