[asterisk-users] PJSIP issues with handling incoming calls

[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 
> <mailto: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)
>> -- 
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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)
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