[asterisk-users] PJSIP issues with handling incoming calls

[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)
> -- 
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