Re: [SR-Users] SIP contact header question

On Wed, Mar 15, 2017 at 11:02:22AM +0100, Daniel-Constantin Mierla wrote: > I haven't followed closely the entire discussion, but for proper > clarification, if one needs to add a header to a received SIP response > that is forwarded, then it has to use append_hf(...) inside on/reply_route. > >

Re: [SR-Users] SIP contact header question

Hello, I haven't followed closely the entire discussion, but for proper clarification, if one needs to add a header to a received SIP response that is forwarded, then it has to use append_hf(...) inside on/reply_route. The append_to_reply() should be used for SIP responses to be generated by

Re: [SR-Users] SIP contact header question

Thanks Alex. And objection duly noted, =) I'll just have to test it out and make sure it doesn't break any of my other clients. Thank you, Jack Davis On Tue, Mar 14, 2017 at 5:46 PM, Alex Balashov wrote: > append_to_reply("Record-Route: \r\n"); > > But I'd like to

Re: [SR-Users] SIP contact header question

append_to_reply("Record-Route: \r\n"); But I'd like to advise against it once more. :) -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users

Re: [SR-Users] SIP contact header question

Thank you for the replies, this has been helpful. So if I do want to go down this road and want to practically implement a solution, what is the best method of appending additional record route headers to a 200 OK response? I guess, specifically, adding them to the 200 OK response that is given

Re: [SR-Users] SIP contact header question

On Tue, Mar 07, 2017 at 03:44:31PM -0600, Jack Davis wrote: > What kind of problems can arise from inserting said header? Theoretically, none. It should be ignored, as per the RFC. Moreover, the RFC doesn't _prohibit_ such a header in an in-dialog request, so you are probably clear to add it.

Re: [SR-Users] SIP contact header question

What kind of problems can arise from inserting said header? I'm trying to find a way to get a pesky setup to start behaving, of course, without requiring them to make any changes. Thank you, Jack Davis On Fri, Mar 3, 2017 at 2:39 PM, Alex Balashov wrote: > On Fri,

Re: [SR-Users] SIP contact header question

On Fri, Mar 03, 2017 at 02:35:10PM -0600, Jack Davis wrote: > Is the preferred method to also include a record-route header in the 200 OK > that is given in response to the re-invite? No. Only initial INVITEs and their replies should have a Record-Route header. In-dialog requests and their

Re: [SR-Users] SIP contact header question

Thank you all for the replies. I'm already utilizing a record-route header in the initial Invite before any re-invites occur. Is the preferred method to also include a record-route header in the 200 OK that is given in response to the re-invite? Thank you, Jack Davis On Fri, Mar 3, 2017 at

Re: [SR-Users] SIP contact header question

For a SIP proxy, the preferred method is indeed to add a Record-Route In the script, add record_route(); instruction. Emmanuel IVèS Le 2017-03-02 à 20:03, Robert Johnson a écrit : I seem to recall reading something stating that you shouldn't modify the contact header - But, I can't find the

Re: [SR-Users] SIP contact header question

No, Kamailio should definitely not be rewriting the Contact. That is set by the remote dialog parties only. If the ACK is being sent directly to the host in the domain portion of the Contact address, that is because the endpoint is ignoring/not honouring Record-Route. That's radioactively wrong

Re: [SR-Users] SIP contact header question

I seem to recall reading something stating that you shouldn't modify the contact header - But, I can't find the text. Might suggest adding a Record-Route header instead of re-writing the contact header? I'm curious as to what the list has to say about this question, I've considered doing

[SR-Users] SIP contact header question

Greetings, I have a general question about the usage of SIP contact headers in the context of using Kamailio as a SIP proxy. [ A ] --> [ Kamailio B ] ---> [ C ] Node A originates a SIP invite, containing a valid via header and URI while setting the