Re: [OpenSIPS-Users] mid_registrar question

2019-09-27 Thread Alexey Vasilyev
Hi Igor, If the phone lost registration, it should also be deleted from mid_registrar. With nathelper, e.g. So next REGISTER should be forwarded, not consumed. But it would be really good solution to have feature like this https://github.com/OpenSIPS/opensips/issues/1769

Re: [OpenSIPS-Users] mid_registrar question

2019-09-27 Thread Igor Olhovskiy
Enpoint B also have 3600 expiry time. So, 1000 + 2600 = 3600. But you got the point. Actually I faced more interesting issue a bit later, maybe actually it was the reason. Only 1 phone involved. Time 0 -> Enpoint A -> mid_registrar -> upstream_registrar (expires 3600) Time 1000 -> Enpoint

Re: [OpenSIPS-Users] mid_registrar question

2019-09-27 Thread Liviu Chircu
Hi, Igor! Correct me if I'm wrong, but doesn't endpoint B think it's registered for another 1000 seconds at step 4) in your example? Anyway, logically speaking, on step 2), the mid-registrar should forward the call to main reg, since there is no guarantee that any of the endpoints will send a

[OpenSIPS-Users] mid_registrar question

2019-09-27 Thread Igor Olhovskiy
Hi! I'm using mid_registrar in mode 2 with default_expires = 3600 Have a following issue. In a case of 2 enpoints registers with expires = 3600 (and actually re-registers in 3599) in a case of 1 phone is down, contact expires on upstream registrar. Like following Time 0 - > Enpoint A ->

Re: [OpenSIPS-Users] Need some clarification in rtpengine module .

2019-09-27 Thread Giovanni Maruzzelli
there is documentation available: https://github.com/sipwise/rtpengine " loop protect Inserts a custom attribute (a=rtpengine:...) into the outgoing SDP to prevent *rtpengine* processing and rewriting the same SDP multiple times. This is useful if your setup involves signalling loops and need

Re: [OpenSIPS-Users] Need some clarification in rtpengine module .

2019-09-27 Thread Sasmita Panda
I am actually not getting how to do this . Somehow add_body_part changing the Content-Type header . Before adding this function : Content-Type: application/sdp After adding this function . Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42 --OSS-unique-boundary-42 Content-Type:

Re: [OpenSIPS-Users] sipmsgops.so / codec_delete_except_re

2019-09-27 Thread Bogdan-Andrei Iancu
Hi Alexey, The logs are quite self-explanatory: DBG:sipmsgops:do_for_all_streams: Message has no SDP Where in the script do you call the codec_delete_except_re() function and for which SIP message ? Please the blow line just before calling codec_delete_except_re(): xlog("--> $rm/$rr,

Re: [OpenSIPS-Users] Problems with loose_route and ACK

2019-09-27 Thread Bogdan-Andrei Iancu
Hi Olle, When doing loose_route(), opensips has to determine if the RURI or the top Route URI points to itself. In order to check this "is for myself", OpenSIPS checks the host URI against the listening IP and the domains from domain table (as the dynamic component). Regards, Bogdan-Andrei

[OpenSIPS-Users] TLS Cypher Renegotiation & WireShark or SIPTrace

2019-09-27 Thread JamesH
I'm trying to use Wireshark to examine my SIP Traffic for an OpenSIPS relay between MS Teams and my SIP Provider. I'm missing AKS from MS Teams and want to look at VIA, RECORD-ROUTE & CONTACT fields in the sip header to check if the correct info is there for the ACK reply to route back The set

Re: [OpenSIPS-Users] codec_delete() not working in 3.0

2019-09-27 Thread Bogdan-Andrei Iancu
Hi MArk, Simply injecting your SDP in a basic script like: route{     # Codecs we do NOT support!     $avp(BadCodec) = "G729";     $avp(BadCodec) = "G729a";     for ($var(ToDelete) in $(avp(BadCodec)[*]))         if (codec_delete("$var(ToDelete)")) {             xlog("CUSTOM_LOG: Invalid

Re: [OpenSIPS-Users] codec_delete() not working in 3.0

2019-09-27 Thread David Villasmil
Annexb as well On Fri, 27 Sep 2019 at 09:36, Mark Farmer wrote: > Thanks James, that's interesting. However my understanding is that all > mention of the target codec should be removed but in this case it seems to > be unchanged. > > According to this list: >

Re: [OpenSIPS-Users] CentOS7: Possible SYN flooding on port 5061

2019-09-27 Thread Alexey Vasilyev
Hi Bogdan, Yes, this is exactly what I need. I'll change it and will see, how it works with next switching. Thanks. - --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html ___ Users

Re: [OpenSIPS-Users] Problems with loose_route and ACK

2019-09-27 Thread Bogdan-Andrei Iancu
Enable log_level 4 (debug level) and post the logs generated by OpenSIPS (172.25.251.4) when handling the ACK. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On

Re: [OpenSIPS-Users] Need some clarification in rtpengine module .

2019-09-27 Thread Callum Guy
I think your syntax is wrong? https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#func_add_body_part On Fri, 27 Sep 2019 at 08:36, Sasmita Panda wrote: > How I will add a attribute in the SDP body . In the rtpengine module there > is no flag which will add a attribute directly . > >

Re: [OpenSIPS-Users] Need some clarification in rtpengine module .

2019-09-27 Thread Sasmita Panda
How I will add a attribute in the SDP body . In the rtpengine module there is no flag which will add a attribute directly . If I am doing like below add_body_part("$var(body)", "a=sdpmangled:yes\r\n"); Its not adding this attribute in the body . Is there any other option to do this ? *Thanks &