Re: [SR-Users] Return codes within a route

Hi, My understanding is that return value of last statement becomes the return value of named route if none explicitly specified. Therefore i always explicitly return 1 or exit. — Muhammad. On Tue 20. Feb 2018 at 22:48, Jeff Bilyk wrote: > Hello, > > Running into odd

Re: [SR-Users] Return codes within a route

Thanks Muhammad, appreciate that clarification. On Tue, Feb 20, 2018 at 5:10 PM, M S wrote: > Hi, > > My understanding is that return value of last statement becomes the return > value of named route if none explicitly specified. Therefore i always > explicitly return 1

Re: [SR-Users] How to Re-transmit the current Request when a negative reply arrives?

On Tue, Feb 20, 2018 at 03:42:03PM +0530, Arun NV wrote: > I am using Kamailio 4.4. And I am forwarding an INVITE to my client. > but sometimes this client will send a negative response.And I am able > to drop that negative response from Kamailio. But I need to > re-transmit the same request to

[SR-Users] How to Re-transmit the current Request when a negative reply arrives?

I am using Kamailio 4.4. And I am forwarding an INVITE to my client. but sometimes this client will send a negative response.And I am able to drop that negative response from Kamailio. But I need to re-transmit the same request to the client again. How do I do that? Thank you, Arunbalan

Re: [SR-Users] RTPengine - Transcoding example available?

Hello! Please explain how to achive next: Side A supports: OPUS PCMA Side B supports: OPUS PCMA PCMU I need to transcode in direction A->B from PCMA to OPUS? So should be like this: A(PCMA) -> (PCMA)Rtpengine(OPUS)->B(OPUS) I configured like this: rtpengine_offer("codec-mask-PCMA

Re: [SR-Users] Kamailio mod ALIAS_DB: multiple aliases

Hey, thanks for the hint with usr_preferences! this is definitively the way to go. Am 20.02.18 um 11:03 schrieb Daniel Tryba: On Tue, Feb 20, 2018 at 09:04:14AM +0100, Stefan R??etschli wrote: So, if the user uses an alias which does not belong to him, i want to overwrite it to the first

Re: [SR-Users] RTPengine - Transcoding example available?

Hi Denys, Am not much user of Kamilio, but am from Telecom background. The reason for your case is, in side A you have OPUS and in side B already OPUS is available and since both has the same codec as first priority it will negotiate itself with same codec and then it will look for other

[SR-Users] Planing next Kamailio releases - v4.4.7 v5.0.6 and v5.1.2

Hello, I am considering to do releases from last three stable branches during next week. v4.4.7 is going to be released out of latest code in branch 4.4 to mark the end of official maintenance for series v4.4.x. The version 4.4.0 was released in March 2016, so it had more or less two years of

[SR-Users] Info: Kamailio World 2018 - Preview With 15 Sessions

Hello, less than 3 months till the start of the 6th edition of Kamailio World Conference, time if flying fast! About one week ago we published the details for a group of accepted speakers, today we made a selection of sessions at the Kamailio World 2018. We had more proposals than we could

Re: [SR-Users] Kamailio mod ALIAS_DB: multiple aliases

On Tue, Feb 20, 2018 at 09:04:14AM +0100, Stefan R??etschli wrote: > So, if the user uses an alias which does not belong to him, i want to > overwrite it to the first alias for his username in the aliases-db. > At the moment, the function alias_db_find returns me the last alias found in > the db.

Re: [SR-Users] Kamailio 4.3 decrypting SIP-TLS

Hello, for Kamailio v4.3 you can print is syslog the $mb as it was suggested in the other response (https://lists.kamailio.org/pipermail/sr-users/2018-February/100378.html), you can do that in request_route or onreply_route. In onsend_route you can print $snd(buf). Then there is siptrace

[SR-Users] Kamailio mod ALIAS_DB: multiple aliases

Hello Kamailio-Users I'm using kamailio 5.1 with Alias_DB module. My subscribers register with a username like 001, 002 etc... Every user has a list of multiple aliases which they can use on their SIP-Trunk. I want to avoid that the user can use any alias for outgoing calls, only the

Re: [SR-Users] RTPengine - Transcoding example available?

Thank you for the answer, but from "SDP" point of view everything looks OK, I think issue is with my config or transcoding engine: Side A INVITE-> Proxy/Rtpengine: v=0 o=- 3543840537 1 IN IP4 s=SIPPER for PhonerLite c=IN IP4 side_a t=0 0 m=audio 55062 RTP/AVP 107 8 a=rtpmap:107 opus/48000/2

Re: [SR-Users] RTPengine - Transcoding example available?

Ok I never tested with kamilio. Probably someone can assist here using kamilio. Thanks. On Tue 20 Feb, 2018, 8:12 PM Denys Pozniak, wrote: > But I use *rtpengine_answer("codec-mask-opus codec-transcode-PCMA")* and > as I understand it should mask opus in OK and

Re: [SR-Users] RTPengine - Transcoding example available?

Yes correct. RTP is good since the OPUS codec is available in both the sides its getting negotiated. Change your config in side A to take only PCMA effective. On Tue, 20 Feb 2018 at 18:05 Denys Pozniak wrote: > Thank you for the answer, but from "SDP" point of view

Re: [SR-Users] RTPengine - Transcoding example available?

But I use *rtpengine_answer("codec-mask-opus codec-transcode-PCMA")* and as I understand it should mask opus in OK and enable transcoding to PCMA. "Change your config in side A to take only PCMA effective." - yes, I tested and it works well, but it is out of my scenario. On 20 February 2018 at

Re: [SR-Users] Inserting a message into a transaction?

sl_send_reply("180", "Make this retarded endpoint ring"); Emmanuel Le 2018-02-20 à 16:13, Daniel Tryba a écrit : For a retarded endpoint (an old ISDN PBX behind a Smartnode) I need to insert a 180 reply before sending a 183 (with SDP), otherwise the PBX fails to playback the inband audio to

Re: [SR-Users] lost mysql connections

Hello juha, Sorry for getting back on this old topic. Can you confirm that with Kamailio v4.3 your problem was solved? Regards, Jan ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org

[SR-Users] Create ISUP Body

Hey Everyone I'm setting up a load testing scenario for my Kamailio. In order to do that i've created another Kamailio to answer "200 OK" to an Invite in order to simulate a call being answered. One of the scenarios I need to test is reading values from the ISUP IAM or ACM body. Here's the

Re: [SR-Users] RTPengine - Transcoding example available?

On 2018-02-20 05:46 AM, Denys Pozniak wrote: Hello! Please explain how to achive next: Side A supports: OPUS PCMA Side B supports: OPUS PCMA PCMU I need to transcode in direction A->B from PCMA to OPUS? So should be like this: A(PCMA) -> (PCMA)Rtpengine(OPUS)->B(OPUS) I configured like

Re: [SR-Users] Inserting a message into a transaction?

On Tue, Feb 20, 2018 at 05:41:49PM +0100, Emmanuel BUU wrote: > sl_send_reply("180", "Make this retarded endpoint ring"); Well, this is embarrassing :) This is so obvious that I didn't even try. I was under the impression that a send_reply in on_reply would send to the source of the reply.

Re: [SR-Users] RTPengine - Transcoding example available?

Thank you, Richard! Works well! On 20 February 2018 at 20:00, Richard Fuchs wrote: > On 2018-02-20 05:46 AM, Denys Pozniak wrote: > >> Hello! >> >> Please explain how to achive next: >> >> Side A supports: >> OPUS >> PCMA >> >> Side B supports: >> OPUS >> PCMA >> PCMU >> >>

[SR-Users] Return codes within a route

Hello, Running into odd behaviour in Kamailio 5.0.2 and I'm hoping to clarify. I have a simple route that uses the textops functions subst and remove_hf. Other than that, there is nothing else done within the route. In some cases, the remove_hf function returns false as there are no matching

[SR-Users] Inserting a message into a transaction?

For a retarded endpoint (an old ISDN PBX behind a Smartnode) I need to insert a 180 reply before sending a 183 (with SDP), otherwise the PBX fails to playback the inband audio to the caller behind the PBX. I can't find hints on how to do this. uac_req_send() can be used to start a new

Re: [SR-Users] Inserting a message into a transaction?

Just a small detail ... I noticed that by doing so, K create a "totag" parameter in the "To" header of 180 Ringing reply. That tag parameter is part of the dialog identification. Then then 183 Session progress that will be forwarded will also have a "totag". It will be different because it

Re: [SR-Users] Kamailio 4.3 decrypting SIP-TLS

Cool, I will check that out. Thank you. :) Robert Remsik Telecom Desk Phone: 970 491 7120 robert.rem...@colostate.edu From: Daniel-Constantin Mierla Sent: Tuesday, February 20, 2018 1:16:59 AM To: Kamailio (SER) - Users Mailing List;