Then the UPDATE either has to follow the Record-Route stack received in 180, or be constructed like the initial INVITE with pre-loaded Route set.
I don't have the time to check the SIP specs right now, but by reusing the Call-Id and From-tag and increasing the CSeq, this is more like a request within dialog, but it doesn't use the To-tag from 180 and no Route headers. So something needs to be fixed inside the UA, the proxy is just following the routing headers that are present in the request.
Cheers,
Daniel
Yes there was a 180 reply from the Callee.
2018/10/11 12:34:57.535510 65.xx.xx.161:64877 -> 65.xx.xx.167:5070SIP/2.0 180 RingingVia: SIP/2.0/UDP 65.xx.xx.167:5070;branch=z9hG4bK98e3.3392254b0b2d98dfc66f12cb7fdba746.0Via: SIP/2.0/UDP 65.xx.xx.172:5060;rport=5060;branch=z9hG4bK694382a1Record-Route: <sip:65.xx.xx.167:5070;r2=on;lr=on;did=bbf.658>Record-Route: <sip:65.xx.xx.167;r2=on;lr=on;did=bbf.658>Contact: <sip:238@10.17.0.35:64877>To: "John"<sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b>;tag=6467b07fFrom: "Robert" <sip:226@mypbx.net>;tag=as0ecef1c4CSeq: 102 INVITEUser-Agent: Bria 4 release 4.8.1 stamp 84929Allow-Events: talk, holdContent-Length: 0
2018/10/11 12:34:57.535631 65.xx.xx.167:5060 -> 65.xx.xx.172:5060SIP/2.0 180 RingingVia: SIP/2.0/UDP 65.xx.xx.172:5060;rport=5060;branch=z9hG4bK694382a1Record-Route: <sip:65.xx.xx.167:5070;r2=on;lr=on;did=bbf.658>Record-Route: <sip:65.xx.xx.167;r2=on;lr=on;did=bbf.658>To: "John"<sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b>;tag=6467b07fFrom: "Robert" <sip:226@mypbx.net>;tag=as0ecef1c4CSeq: 102 INVITEAllow-Events: talk, holdContent-Length: 0
On Mon, Oct 15, 2018 at 12:15 PM Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,
indeed, I was misled by the Route headers in INVITE, which looked like inside a dialog, but the parameter in To header is rinstance.
Is there any 18x response?
Cheers,
Daniel
On 15.10.18 16:00, Sergiu Pojoga wrote:
Hi again,
Hmm... I don't see a To-tag in the INVITE, neither there's a 200OK to provide because the UPDATE was sent out prior to the callee answering the call.
If there should be a Route header in the UPDATE, it would it indicate a bug with Asterisk firing off the UPDATE without a pre-set Route dictated by the Path?
If that's the case, I suppose my options are:
- reach out to Asterisk to investigate and fix it (unrealistic)
- store the Route header from the initial INVITE in a AVP and insert it later if an UPDATE follows. Would that break anything up?
Any other constructive suggestions?
Thanks.
On Mon, Oct 15, 2018 at 2:34 AM Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,
that seems to be a re-INVITE (has To-tag). I would need at least the initial INVITE and the 200ok, along with the UPDATE request.
If the UPDATE is after the re-INVITE, it is missing the Route header as in the re-INVITE.
Cheers,
Daniel
On 12.10.18 16:53, Sergiu Pojoga wrote:
Hi Daniel,
Certainly, below find the initial INVITE and the subsequent UPDATE, as received by Kamailio@65.xx.xx.167. If those aren't sufficient, let me know and if it's ok with you, I'll send the full pcap in private.
The dilemma in my mind is whether the UPDATE should have a pre-set Route header, similar to how the INVITE has.
2018/10/11 12:34:57.339306 65.xx.xx.172:5060 -> 65.xx.xx.167:5060INVITE sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b SIP/2.0Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK694382a1Max-Forwards: 70Route: <sip:65.xx.xx.167;lr;received=sip:65.xx.xx.161:64877;r2=on>,<sip:xx.xx.xx.167:5070;lr;received=sip:65.xx.xx.161:64877;r2=on>From: "Robert" <sip:226@mypbx.net>;tag=as0ecef1c4Contact: <sip:226@65.xx.xx.172:5060>CSeq: 102 INVITE Supported: replaces, timer, pathContent-Type: application/sdpContent-Length: 386
2018/10/11 12:35:06.096457 65.xx.xx.172:5060 -> 65.xx.xx.167:5060UPDATE sip:238@10.17.0.35:64877;alias=65.xx.xx.161~64877~1 SIP/2.0Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK34fab05cMax-Forwards: 70From: "Robert" <sip:226@mypbx.net>;tag=as0ecef1c4To: <sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b>;tag=6467b07fContact: <sip:226@65.xx.xx.172:5060>CSeq: 103 UPDATEContent-Length: 0
Much obliged.
On Fri, Oct 12, 2018 at 9:38 AM Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,
you hve to provide the sip traffic for this case, the screenshot doesn't show the sip headers used for routing in this case, therefore grab the sip traffic for all sip messages in such scenarion, either ngrep output or pcap file, and send it over to see if some headers are missing or not set properly.
Cheers,
Daniel
On 11.10.18 21:03, Sergiu Pojoga wrote:
Hi ppl,
I have this problem with call transfer, may be someone can help.
The phone to the far right is registered with the Registrar to the far left using two PATH headers (trespassing two proxy ports, 5070 then 5060).
As you can see in the graph below, after receiving the UPDATE request, Kamailio relays it further from port 5060, I expect it to be from 5070 just like the dialog forming INVITE and the CANCEL afterwards.
The UPDATE has a to-tag, but unlike the original INVITE - it has no Route header!???
route[WITHINDLG] {
if (!has_totag()) return;if (loose_route()) {
route(DLGURI);if (is_method("BYE")) {...}
else if ( is_method("ACK") ) {route(NATMANAGE);}else if ( is_method("NOTIFY") ) {record_route();}
route(RELAY);exit;}
if ( is_method("ACK") ) {...}
# handle UPDATE method for in-dialog requestsif (is_method("UPDATE")) {route(DLGURI);record_route();route(RELAY);}}
Thanks in advance.
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-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com