Re: [SR-Users] ims_dialog default_timeout parameter

Hi, Daniel No, the dialogs were not terminated after 10 minutes if no BYE message was sent. But if Default value(43200) was set, the S-CSCF will run out of memory finally (after 20 calls). Have any relation with this parameter and memory release ? Any impacts if using value 600 at

[SR-Users] no corresponding socket found for

Hi I have a setup with three kamailio's in the same host but on different ports. client ->5069 -> 5070 -> 5071 -> SIP provider Now I get this error from the 5069 Kamailio when it is routing a 200 Ok to the client: Apr  9 21:54:10 kfcpbx /usr/sbin/kamailio[9688]: ERROR: [forward.c:181]:

[SR-Users] rtpengine no voice when working between DTLS-SRTP endpoints

Greetings list, I am trying to make two endpoints talking on DTLS-SRTP. But I hear on audio. Things work perfectly fine if I use RTP or SRTP with TLS. Endpoints are pjsip based application not webrtc based clients. Below are logs from rtpengine. I hope someone could point out amiss. Apr 9

Re: [SR-Users] rtpengine installation dependencies

So, many people to help you out. Such an amazing community we have. Thanks, everyone. Br, Aqs. On 9 April 2018 at 19:30, Igor Olhovskiy wrote: > Hi! > rtpengine can be installed for deb8 from repos > > First you need to install mariadb 10 (smth like

Re: [SR-Users] Calling a mobile SIP client, which needs to update the registration

Perhaps this presentation can be of inspiration. You would need to branch Your call if You have more destinations. A simpler approach which I use is to loop for some seconds after sending the push, and see if the registration shows up in the location table. Med

Re: [SR-Users] ims_dialog default_timeout parameter

Hello, that should be the call lifetime. Are the dialogs terminated after 10 minutes? Cheers, Daniel On 09.04.18 16:58, tyd wrote: > Dear all, > > What's the meaning of ims_dialog  default_timeout parameter ? > /Default value is 「43200 (12 hours)」/ > > When modparam("ims_dialog",

[SR-Users] ims_dialog default_timeout parameter

Dear all, What's the meaning of ims_dialog default_timeout parameter ? *Default value is 「43200 (12 hours)」* When modparam("ims_dialog", "default_timeout", 600) was set, the situation of Kamailio S-CSCF out of memory was solved and is running smoothly. Debian 8 with Kamailio 4.4.7

Re: [SR-Users] rtpengine installation dependencies

Hi! rtpengine can be installed for deb8 from repos First you need to install mariadb 10 (smth like  https://www.tecmint.com/install-mariadb-in-debian/) next - you can use ng-voice repos for rtpengine http://repository.ng-voice.com/ # Key: # wget -O - http://repository.ng-voice.com/PublicKey |

Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive delivery

Dear team, Thank you for your reply. @MS & @ Davld, Let me test again and share traces. @Zaka, I will check the save() result as you suggested. Location is being saved and properly looked up because messages are being delivered (and not stored in DB ) when both parties are online. -- regards,

Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive delivery

Please check the return value of save and replace the logic: - $var(save_result) = save("location") ; if ($var(save_result) <= -1) { xlogl("L_ERROR", "[$rm] -> Unable to save user location \n"); sl_reply_error(); exit(); } if ($var(save_result) == 1) { xlogl("L_INFO", "[$rm] ->

Re: [SR-Users] rtpengine

But if side A has G729, probably you can avoid transcoding by stripping all codecs and offer needed. *rtpengine_offer("codec-strip-all codec-offer-G729")* 2018-04-09 12:47 GMT+02:00 Denys Pozniak : > I think you need to mask all codecs from A side and transcode g729,

Re: [SR-Users] rtpengine

I think you need to mask all codecs from A side and transcode g729, like this: *rtpengine_offer("codec-mask-all codec-transcode-G729")* 2018-04-09 12:35 GMT+02:00 : > Hello Denys, > I am trying remove all codec and offer only G729A. > Based on doc I can do some think

Re: [SR-Users] rtpengine

Hello Denys, I am trying remove all codec and offer only G729A. Based on doc I can do some think like this ? codec-offer-G729 Slava On Mon, Apr 9, 2018 at 5:05 AM, Denys Pozniak wrote: Hello! https://github.com/sipwise/rtpengine/blob/master/README.md You can use

Re: [SR-Users] rtpengine installation dependencies

Hello! Try to use backports repo https://packages.debian.org/hu/jessie-backports/debhelper 2018-04-07 0:58 GMT+02:00 Sergey Okhapkin : > Please do not reply to the same mailing list (with 2 different names > though) > twice. One of aliases

Re: [SR-Users] rtpengine

Hello! https://github.com/sipwise/rtpengine/blob/master/README.md You can use options like codec-transcode-... (dash instead of equals). 2018-04-09 1:40 GMT+02:00 : > Hello Everyone, > What flags available on latest rtpegnine to acomplish transcoding and > codec

Re: [SR-Users] Calling a mobile SIP client, which needs to update the registration

Look at https://www.kamailio.org/docs/modules/5.0.x/modules/tsilo.html Regards, Igor On Apr 9, 2018, 9:44 AM +0300, Ulrich Henning , wrote: > Hi everybody, > > I am trying to call a sip client on a mobile cellphone, which is registered > fine at my Kamailio instance.

[SR-Users] Calling a mobile SIP client, which needs to update the registration

Hi everybody, I am trying to call a sip client on a mobile cellphone, which is registered fine at my Kamailio instance. Everything is working fine if the phone is awake and the mobile app is not sleeping (e.g. energy saving by OS). If the device is sleeping, the cellphone gets a wakeup request