Re: [OpenSIPS-Users] Async event_route and cachdb_redis

Thanks Razvan for dedicating time for me. You can find the output from the given command here: http://pastebin.com/fh11mkXS On Wed, Jun 22, 2016 at 12:48 PM, Răzvan Crainea wrote: > Could you run the 'opensipsctl trap' command and paste the output on > pastebin. > > Best

Re: [OpenSIPS-Users] CURL library with respect to REST_API calls

Hi team, Any one has any clue on the below topic? Regards, Agalya From: Ramachandran, Agalya (Contractor) Sent: Monday, June 20, 2016 1:45 PM To: OpenSIPS users mailling list Subject: CURL library with respect to REST_API calls Hi team, I have a question regarding

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

Could you run the 'opensipsctl trap' command and paste the output on pastebin. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/22/2016 07:39 PM, SamyGo wrote: Yes correct. Async event route even stops to be executed. On Jun 22, 2016 12:37, "Răzvan Crainea"

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

Yes correct. Async event route even stops to be executed. On Jun 22, 2016 12:37, "Răzvan Crainea" wrote: > So the patch doesn't do anything but stops triggering the event? > > Regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/22/2016 07:07

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

So the patch doesn't do anything but stops triggering the event? Regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/22/2016 07:07 PM, SamyGo wrote: Yeah it only happens at startup. If I start opensips in debug_mode=yes then the error prints for infinite time. With

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

Yeah it only happens at startup. If I start opensips in debug_mode=yes then the error prints for infinite time. With your patch; putting "async" doesn't even call the event route. If I remove async attribute then it works just like before the patch. Regards, Sammy On Wed, Jun 22, 2016 at 3:10

Re: [OpenSIPS-Users] DNS-SRV query in opensips

Hi, They are different. RURI is the part in request's first line, while the DURI is a an outbound proxy used just to finding the destination at network level (it will not be present in the SIP request). by using the set functions you do not create a new branch, you are just changing the

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

Hi Răzvan Crainea. Thank you very much for trying to help me. Yesterday my boss asked me to work in another part of our project. So, I will have to pause this verification for a while. When I return to it, I will check the log. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence

Re: [OpenSIPS-Users] Leak AVPOS + SQLITE

Ok. Thank you very much! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org em nome de Razvan Crainea Enviado:

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

Ben is correct. In my opinion, a very easy solution. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Newlin, Ben Sent: Tuesday, June 21, 2016 5:24 PM To: sevpal ; OpenSIPS users mailling list Subject:

Re: [OpenSIPS-Users] webrtc native client for opensips

OK but in this i will not have control over webrtc codecs. What I am looking for is ... 1- Native application can send SIP messages over wss. 2- For media instead of using chrome functions , use my custom webrtc ( https://webrtc.org/native-code/ios/) which I use as library I am not even sure at

Re: [OpenSIPS-Users] webrtc native client for opensips

If you intend on running on android and iOS restcomm appears to have native clients that support sip over websocket . I've tested the iOS app they with OpenSIPS and baseline functionality was there . https://github.com/RestComm/restcomm-android-sdk On Jun 22, 2016 5:34 AM, "John Nash"

Re: [OpenSIPS-Users] webrtc native client for opensips

My objective is to make a native webrtc application which can use SIP over wss for signalling and for media also I do not want to be dependent on chrome as in future I wish to incorporate more codecs into it. Any pointers for me? On Wed, Jun 22, 2016 at 2:55 PM, Tito Cumpen

Re: [OpenSIPS-Users] webrtc native client for opensips

John, You can utilize sipjs and jssip on account that they utilize sip over websocket. Take into consideration that chrome will only allow getusermedia if you are using wss and https . On Jun 22, 2016 3:07 AM, "John Nash" wrote: > Apart from sipml5 is there any native

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

It also seems like AVPOPS module [1] may be a good solution here as it has functions to pull data from a database into AVPs based by user. [1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html Ben Newlin From: on behalf of sevpal

Re: [OpenSIPS-Users] B2BUA not responding to reinvites

After some stracing of opensips I found that the reinvites where not making it to opensips then tracked the issue down to fail2ban :/ solution: aptitude remove ufw fail2ban I am going back to hand crafted iptables rules [] thanks for the help. Kevin

Re: [OpenSIPS-Users] DNS-SRV query in opensips

Hi Agalya, seturi() and setdsturi() set the RURI / DestinationURI for the current message / branch. When youdo an append_branch() a new branch is stored for serial/parallel forking (note that the current branch does not changes - this is branch number 1). So, append_branch() will make a

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

Hi, Sammy! Does this happen only at startime, or happens during runtime too? Regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/21/2016 10:57 PM, SamyGo wrote: Hi , After recompiling , when I start opensips it gives this error:

[OpenSIPS-Users] webrtc native client for opensips

Apart from sipml5 is there any native webrtc client also which I can explore to work with opensips? The examples I find for webrtc native seem to be using jingle protocol but in case to make it work with opensips, It has to use SIP/SDP at client end right?..Any examples?

Re: [OpenSIPS-Users] Leak AVPOS + SQLITE

Hi, Rodrigo! Valgrind may report some memory allocated, and not freed, but that is not necessarily a memory leak. There is a single block of 1024 bytes not freed during runtime, so I think that is peanuts. The memory used by OpenSIPS is not allocated with malloc, so cannot be traced by

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

Hi, Rodrigo! Can you print the $ru variable before and after each lookup() query? Something like: $var(ru) = $ru; xlog("R-URI before caller lookup: $ru\n"); lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru =