[SR-Users] Dispatcher Module & ds_load_list enhancement

Just an observation and enhancement suggestion. VERY occasionally Ive been getting *ERROR: [parser/parse_param.c:591]: parse_params2(): parse_params(): No memory left* Now I know this is the root of the problem BUT ... it would be nice if it didnt leave my kamailio with no dispatchers at all.

[SR-Users] Asynchronous processing increases branch count

Hi, Normally with serial forking, the first branch ID corresponds to $T_branch_idx == 0, i.e. Via: SIP/2.0/UDP 172.30.110.4;branch=z9hG4bK7f3d.57593ee3a139f0ae6c4a8a0f531e342c.0 In the event of failure, a second branch attempt has a branch index of 1 and so on: Via: SIP/2.0/UDP 172.30.1

Re: [SR-Users] Issue with BYE and $du

Hi Rene, On 12/10/2015 06:01 PM, Rene Montilva wrote: What could be the scenario, when hangup a call and kamailio show empty the $du var and can't send BYE to endpoint $du is ordinarily only set when the network and transport-layer destination of the request is different to where consuming t

[SR-Users] Issue with BYE and $du

Hi list What could be the scenario, when hangup a call and kamailio show empty the $du var and can't send BYE to endpoint -- Ing. Rene Montilva *FOSS Developer and VoIP Engineer.* ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing

[SR-Users] transformations {s.rm,match}

Hi Guys, I have been using {s.rm,match} to remove occurrences of numbers, however I am aware it removes all occurrences. Is it possible to make it only remove the first occurrence and not all? For example If I am looking to modify 639157407639 I want to remove only the first 63, not both. So I

[SR-Users] problem with cfgtrace

Hi, Have just installed kamailio 4.3.4 and for some reason not getting cfgtraces from debugger. I've picked the default kamailio-advanced.cfg, and did no changes other than setting #!define WITH_DEBUG. Any suggestions on what could be causing this to happen? Thanks! Joao Alves This message

Re: [SR-Users] WebRTC no longer supports RTP

I already use DTLS-SRTP (websockets dont works with RTP). This is my SDP body. And I have no sound at incoming calls tcpdump shows me that I have no rtp strean fro websocket endpoint v=0 o=root 1828066564 1828066564 IN IP4 1.1.1.1 s=Cattaxi Media Server c=IN IP4 1.1.1.1 t=0 0 m=audio 30328 RTP/SA

Re: [SR-Users] WebRTC no longer supports RTP

Hi! use DTLS-SRTP, to say how to handle it with rtpengine - I think you should provide more info about your setup, and call cases Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/WebRTC-no-longer-supports-RTP-tp143834p143836.html Sent from the Users mailing list

Re: [SR-Users] info: kamailio.org updates - letsencrypt for https and dkim for email

Am 08.12.2015 um 09:06 schrieb Daniel-Constantin Mierla: > Also, if anyone has more hints on increasing the security/privacy for > the web server and email systems we run for kamailio.org, do not > hesitate to provide us suggestions. Create a permanent redirect from the HTTP websites to the HTTPS

[SR-Users] WebRTC no longer supports RTP

https://developers.google.com/web/updates/2015/10/chrome-47-webrtc So at 47 chrome we already have no sound. What kind of proto we must use and how to handle this with rtpengine? Do anyone have same problems with it? ___ SIP Express Router (SER) and Kama

Re: [SR-Users] rtpengine and serial forking with multiple network interfaces

I tested an approach in which all initial rtpengine_manage() calls (in support of a new SDP offer) were made in a branch_route[], and this appeared to work. Is this the 'orthodox' way to do it? -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, G

[SR-Users] rtpengine and serial forking with multiple network interfaces

Hi, It is not very clear to me how to handle serial forking scenarios with rtpengine where there are multiple RTP listeners attached to multiple network interfaces. That is to say, given: rtpengine ... -i net1/xxx.xxx.xxx.xxx -i net2/yyy.yyy.yyy.yyy 1. I attempt to reach SIP gateway 1 vi

Re: [SR-Users] RTPPROXY unpredictable behavior

Hi Javohir, don't worry, your english is fine. Can you send us the parameters you have set for rtpproxy in your kamailio cfg? You should find something like this: modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:2") Make sure it matches the configuration of the RTPProxy. Thanks, Carsten

Re: [SR-Users] having a count of lost response for dialog keep alives

Thanks Daniel. Have opened an issue related to this. I cannot label though !! https://github.com/kamailio/kamailio/issues/438 Please label it as a Feature Request. Thanks. - Jayesh On Thu, Dec 10, 2015 at 1:44 PM Daniel-Constantin Mierla wrote: > Hello, > > On 10/12/15 08:25, Jayesh Nambiar wro

Re: [SR-Users] issue with BYE

Hello, you need to provide full sip trace from initial invite to the bye in order to see if some routing headers are not carrying the proper values. You can use: ngrep -d any -qt -W byline "sip" port 5060 Do such call exposing the issue and sent the network sip traffic here. Cheers, Daniel On

Re: [SR-Users] random incorrect method parsing

Hi Daniel! just additional info about the nature of that "extra zero". Vendor use them as a padding bytes RFC http://www.ietf.org/rfc/rfc0894.txt says: "If necessary, the data field should be padded (with octets of zero) to meet the Ethernet minimum frame size. This padding is not part of

Re: [SR-Users] having a count of lost response for dialog keep alives

Hello, On 10/12/15 08:25, Jayesh Nambiar wrote: > Hi, > I'm using dlg_set_property(ka-src) and dlg_set_property(ka-dst) to > keep alive my clients. In case of a network change the client does a > Re-Invite with the new contact address and kamailio does keep-alives > to the new address properly as

Re: [SR-Users] random incorrect method parsing

Hello, update on this -- I pushed last evening to master a slightly different patch, to skip other white space characters before the method is started to be parsed. Not having much time, I didn't dig further in the code, but the position of the patch suggests that these leading whitespaces are st

Re: [SR-Users] [Kamailio] Perfomance tests

Hello, no updates were published for those tests. Anyhow, same scenarios should be easy to do with the latest version. If I get some time in the near future, I will try to update them. If you search on the archive, from time to time there were people publishing details about their tests with thei

[SR-Users] RTPPROXY unpredictable behavior

I am trying to enable RTPPROXY on debian, but it seems RTPProxy is ignoring my arguments... So, I am launching RTPPROXY using command: rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -p /var/run/rtpproxy.pid -R -a -P -r /tmp/rtppath -S /tmp/rtpspool -u rtpproxy rtpproxy Also tried with: rtppro