[asterisk-users] PJSIP issues with handling incoming calls

[Rainer Piper]

PS all incoming calls are directed to sipgatefilter in extentions.conf 
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just 
look at the cli output *NoOp(**** 49${gotoadr:-11} ****)

Am 02.09.2014 um 17:04 schrieb Rainer Piper:
> 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)
>
>


-- 
*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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