[OpenSIPS-Users] t_on_reply not catching 183

Mark Farmer farmorg at gmail.com
Mon Jun 10 09:18:00 EDT 2019


Thanks. The soft client is registered to an Asterisk box so I don't think
that's the issue here.



On Sat, 8 Jun 2019 at 00:58, Schneur Rosenberg <rosenberg11219 at gmail.com>
wrote:

> Are you missing a fix_natted_contact somewhere? That should in most cases
> eliminate the need of  stun.
>
> On Fri, Jun 7, 2019, 2:14 PM Mark Farmer <farmorg at gmail.com> wrote:
>
>> Never mind, it seems the issue was triggered by my soft client not using
>> STUN
>>
>> Fixed that bit now :)
>>
>> Mark.
>>
>>
>> On Fri, 7 Jun 2019 at 10:53, Mark Farmer <farmorg at gmail.com> wrote:
>>
>>> Sorry - version info:
>>>
>>> opensips -V
>>> version: opensips 2.4.5 (x86_64/linux)
>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>> poll method support: poll, epoll, sigio_rt, select.
>>> git revision: d025b4f61
>>> main.c compiled on 11:39:27 Apr 12 2019 with gcc 7
>>>
>>>
>>> On Fri, 7 Jun 2019 at 10:45, Mark Farmer <farmorg at gmail.com> wrote:
>>>
>>>> Hi everyone, I'm trying to solve an issue related to reception of early
>>>> media. I am getting 183 messages back to establish the audio path but
>>>> OpenSIPS is passing the 183 back to an Asterisk box without changing the
>>>> SDP "c" parameter. So Asterisk tries to send audio direct instead of to the
>>>> rtpproxy.
>>>>
>>>> Doing some debugging, I can see that my reply route doesn't seem to be
>>>> matching the 183 because I never get any logs from the reply route. Can
>>>> anyone see anything wrong with it? I've checked the routing script & the
>>>> reply route seems to be armed.
>>>>
>>>> onreply_route[DROUTING] {
>>>>
>>>>         if (is_method("BYE|CANCEL")) {
>>>>                 sip_trace("htid","d");
>>>>                 rtpproxy_unforce();
>>>>         }
>>>>
>>>>         #if ( $rs >= 200 )
>>>>                 #$acc_extra(to_usr) = $tU;
>>>>
>>>>         if ($rs=~"(2[0-9][0-9])|(183)" && has_body("application/sdp")) {
>>>>             xlog("Processing reply $fU");
>>>>             if (is_from_gw("1")) {
>>>>                     xlog("Reply from Asterisk PBX");
>>>>                     setflag(INT_R);
>>>>                 } else if (is_from_gw("2")) {
>>>>                         xlog("Reply from Provider");
>>>>                         setflag(EXT_R);
>>>>                 }
>>>>         }
>>>>
>>>>         if (isflagset(INT_R)) {
>>>>                 remove_hf("P-Asserted-Identity");
>>>>                 rtpproxy_answer("corwfei");
>>>>             } else if (isflagset(EXT_R)) {
>>>>                 rtpproxy_answer("corwfie");
>>>>         }
>>>> }
>>>>
>>>>
>>>> Many thanks for any & all help.
>>>> Mark.
>>>>
>>>>
>>>
>>> --
>>> Mark Farmer
>>> farmorg at gmail.com
>>>
>>
>>
>> --
>> Mark Farmer
>> farmorg at gmail.com
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
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>>
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-- 
Mark Farmer
farmorg at gmail.com
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