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 <[email protected]> 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:[email protected]:52666;rinstance=d4284982f7c18786>. > To: <sip:[email protected]>. > From: <sip:[email protected]>;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:[email protected];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:[email protected]:35631;transport=UDP>. > To: <sip:[email protected];transport=UDP>. > From: <sip:[email protected];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:[email protected]: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:[email protected]:5070>;tag=as3daea415. > To: <sip:[email protected]:52666;rinstance=d4284982f7c18786>. > Contact: <sip:[email protected]:5070>. > Call-ID: [email protected]: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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