[asterisk-users] PJSIP issues with handling incoming calls

[Nick Awesome]

Ok, thanks for an answer. That solution works.

On 02 Sep 2014, at 22:36, Rainer Piper <rainer.piper at soho-piper.de> wrote:

> contact_user in pjsip.conf has to point to the filter or to an agi in the extentions.conf
> like:
> 
> pjsip.conf
> contact_user=blablabla
> 
> extensions.conf
> exten => blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
> 
> 
> Am 02.09.2014 um 20:11 schrieb Rainer Piper:
>> contact_user can be anything and calling an agi is no problem 
>> 
>> 
>> Am 02.09.2014 um 19:49 schrieb Nick Awesome:
>>> Okay, contact_user seems like do the job. Thanks
>>> is contact_user can be anything, or it should be same as username ?
>>> I would like to use contact_user for transmitting trunk name into agi script
>>> 
>>> On Sep 2, 2014, at 7:04 PM, Rainer Piper <rainer.piper at soho-piper.de> wrote:
>>> 
>>>> I use in pjsip.conf 
>>>> [sipgate1]
>>>> type=registration
>>>> transport=transport-udp
>>>> outbound_auth=sipgate1_auth
>>>> server_uri=sip:sipgate.de
>>>> client_uri=sip:555123456 at sipgate.de
>>>> contact_user=sipgatefilter ; goto the filter in extensions.conf
>>>> retry_interval=60
>>>> forbidden_retry_interval=600
>>>> expiration=3600
>>>> 
>>>> extensions.conf ; i'm cutting the dialed number out of the invite Header and goto/jump to the extensions
>>>> ; incoming VOIP 9716716x SIPGATE
>>>> exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)} ***)
>>>>     same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
>>>>     same => n,NoOp(**** 49${gotoadr:-11} ****)
>>>>     same => n,Goto(49${gotoadr:-11},1)
>>>> 
>>>> ; the filter is jumping to the extensions ...
>>>> 
>>>> ; incoming VOIP 97167160 SIPGATE -> MENU
>>>> exten => 4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)
>>>> ; incoming VOIP 97167161 SIPGATE
>>>> exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
>>>> ; incoming VOIP 97167162 SIPGATE ECHO TEST
>>>> exten => 4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167163 SIPGATE
>>>> exten => 4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167164 SIPGATE
>>>> exten => 4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167165 SIPGATE
>>>> exten => 4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incncoming VOIP 97167166 Mailbox
>>>> exten => 4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167167 CONF. 1
>>>> exten => 4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167168 CONF. 2
>>>> ;exten => 4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> exten => 4922897167168,1,Answer
>>>>         same => n,echo()
>>>>         same => n,Hangup()
>>>> ; incoming VOIP 97167169 FAX
>>>> ;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> 
>>>> 
>>>> Regards
>>>> Rainer
>>>> 
>>>> Am 02.09.2014 um 15:08 schrieb Joshua Colp:
>>>>> Nick Awesome wrote: 
>>>>>> register =>  73432260005:pass at 10001 
>>>>>> register =>  73432260050:pass at 10002 
>>>>>> 
>>>>>> [10001] 
>>>>>> type=peer 
>>>>>> host=80.75.132.66 
>>>>>> context=dialmap 
>>>>>> [10002] 
>>>>>> type=peer 
>>>>>> host=80.75.132.66 
>>>>>> context=dialmap 
>>>>> 
>>>>> Can you provide a sip debug of calls to both of these? I'm confused how that... works... 
>>>>> 
>>>> 
>>>> 
>>>> -- 
>>>> Rainer Piper 
>>>> Integration engineer 
>>>> Koeslinstr. 56 
>>>> 53123 BONN 
>>>> GERMANY 
>>>> Phone: +49 228 97167161 
>>>> P2P: sip:7000 at sip.soho-piper.de:5072 (pjsip-test)
>>>> -- 
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>> 
>>> 
>>> 
>> 
>> 
>> -- 
>> Rainer Piper 
>> Integration engineer 
>> Koeslinstr. 56 
>> 53123 BONN 
>> GERMANY 
>> Phone: +49 228 97167161 
>> P2P: sip:7000 at sip.soho-piper.de:5072 (pjsip-test)
>> 
>> 
> 
> 
> -- 
> Rainer Piper 
> Integration engineer 
> Koeslinstr. 56 
> 53123 BONN 
> GERMANY 
> Phone: +49 228 97167161 
> P2P: sip:7000 at sip.soho-piper.de:5072 (pjsip-test)
> -- 
> _____________________________________________________________________
> -- 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

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