[asterisk-users] Getting Asterisk to use the SIP Path header

[Peter Baines]

If anyone else is having this issue, asterisk 1.8.32.3 uses the Path header
as expected, if you want to follow progress I've created a bug report:
https://issues.asterisk.org/jira/browse/ASTERISK-25666


On 6 January 2016 at 10:29, Peter Baines <lists at pbaines.com> wrote:

> Hi,
>
> How do I get asterisk to use the SIP Path header value from registrations
> when calling devices?
>
> I am trying to use opensips as a proxy for asterisk, when a client
> registers I am adding the Path header before forwarding the REGISTER onto
> asterisk. The problem is when asterisk recieves an INVITE it does not use
> the value from the Path header, it is sending directly to the device. Can
> anyone point me in the right direction as to why?
>
> I am using asterisk 13.6.0 with the default configuration, the changes I
> have made are:
>
> In sip.conf I have uncommented:
> supportpath=yes
> rtsavepath=yes
>
> In users.conf I have:
>
> [6000]
> secret =
> host=dynamic
> context = default
>
> [6001]
> secret =
> host=dynamic
> context = default
>
> [6002]
> secret =
> host=dynamic
> context = default
>
> In extensions.conf I have made default like:
> [default]
> ;include => demo
> exten => 6000,1,Dial(SIP/6000,18)
> exten => 6000,n,Hangup()
>
> exten => 6002,1,Dial(SIP/6002,18)
> exten => 6002,n,Hangup()
>
> exten => 6001,1,Dial(SIP/6001,18)
> exten => 6001,n,Hangup()
>
>
> Below is the 6000 user REGISTER going from opensips (10.15.20.137:5060)
> into asterisk (192.168.68.68:5070) with the Path header.
>
> U 2016/01/06 10:04:23.399170 10.15.20.137:5060 -> 192.168.68.68:5070
> REGISTER sip:10.15.20.137 SIP/2.0.
> Via: SIP/2.0/UDP 10.15.20.137:5060;branch=z9hG4bKcc2c.b40fb511.0.
> Via: SIP/2.0/UDP 10.15.20.53:52666
> ;received=10.15.20.53;branch=z9hG4bK-d8754z-91422161f08a7943-1---d8754z-;rport=52666.
> Max-Forwards: 69.
> Contact: <sip:6000 at 10.15.20.53:52666;rinstance=d4284982f7c18786>.
> To: <sip:6000 at 10.15.20.137>.
> From: <sip:6000 at 10.15.20.137>;tag=9e95da50.
> Call-ID: OTQ1ZTdmZmE3OTM1ZWVkYzMzYWZiMDMzMDgyODhmOTU.
> CSeq: 2 REGISTER.
> Expires: 3600.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> User-Agent: Bria 3 release 3.5.5 stamp 71243.
> Content-Length: 0.
>
>
>
> *Path: <sip:10.15.20.137;lr>.*
> Below is the INVITE going from opensips to asterisk for 6000
>
> U 2016/01/06 10:11:13.668929 10.15.20.137:5060 -> 192.168.68.68:5070
> INVITE sip:6000 at 10.15.20.137;transport=UDP SIP/2.0.
> Record-Route: <sip:10.15.20.137;lr;nat=yes>.
> Via: SIP/2.0/UDP 10.15.20.137:5060;branch=z9hG4bK8f77.9d6ef7e7.0.
> Via: SIP/2.0/UDP 188.39.51.2:35631
> ;rport=35631;received=10.15.20.53;branch=z9hG4bK-d8754z-d46f3a0333dc5d49-1---d8754z-.
> Max-Forwards: 69.
> Contact: <sip:6001 at 10.15.20.53:35631;transport=UDP>.
> To: <sip:6000 at 10.15.20.137;transport=UDP>.
> From: <sip:6001 at 10.15.20.137;transport=UDP>;tag=870fdf72.
> Call-ID: YWVjN2VjMDZmYmZmNjg4MTE2MzJlZGU1ZDNjZGU2NDc..
> CSeq: 2 INVITE.
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE.
> Content-Type: application/sdp.
> Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri.
> User-Agent: Z 3.3.21933 r21903.
> Allow-Events: presence, kpml.
> Content-Length: 237.
> .
> v=0.
> o=Z 0 0 IN IP4 188.39.51.2.
> s=Z.
> c=IN IP4 188.39.51.2.
> t=0 0.
> m=audio 8000 RTP/AVP 3 110 8 0 98 101.
> a=rtpmap:110 speex/8000.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode=20.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
>
>
> I would now expect asterisk to send the INVITE to the value of the Path
> header in the registration (10.15.20.137:5060) however it is sending the
> INVITE directly to the device (10.15.20.53:52666):
>
> U 2016/01/06 10:11:13.671345 192.168.68.68:5070 -> *10.15.20.53:52666
> <http://10.15.20.53:52666>*
> INVITE sip:6000 at 10.15.20.53:52666;rinstance=d4284982f7c18786 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.68.68:5070;branch=z9hG4bK308a4ef5;rport.
> Max-Forwards: 70.
> Route: <sip:10.15.20.137;lr>.
> From: "New User" <sip:6001 at 192.168.68.68:5070>;tag=as3daea415.
> To: <sip:6000 at 10.15.20.53:52666;rinstance=d4284982f7c18786>.
> Contact: <sip:6001 at 192.168.68.68:5070>.
> Call-ID: 55202bc71f9e684d0b82c7cb2e8684ab at 192.168.68.68:5070.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 13.6.0.
> Date: Wed, 06 Jan 2016 10:11:13 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces, timer, path.
> Content-Type: application/sdp.
> Content-Length: 286.
> .
> v=0.
> o=root 887525354 887525354 IN IP4 192.168.68.68.
> s=Asterisk PBX 13.6.0.
> c=IN IP4 192.168.68.68.
> t=0 0.
> m=audio 12356 RTP/AVP 0 8 3 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=maxptime:150.
> a=sendrecv.
>
>
> Thanks,
> Peter
>
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