[SR-Users] RTP Engine multiplexing and De-multiplexing

Hello, A call from a remote webrtc client is coming to (opensips+rtpengine). The media streams from the webrtc client is multiplexed. Can I use rtpengine to demultiplex the multiplexed streams and send it to other end as de-multiplexed SRTP traffic . This is because, the other end handle the srtp

Re: [SR-Users] rtpengine graphite statistics server

Yes, the Graphite server is openly available: http://graphite.wikidot.com/ Greetings, Paul Am 13.01.2016 um 03:53 schrieb Juha Heinanen: rtpengine README.md mentions "graphite statistics server". is that server openly available somewhere? -- juha

[SR-Users] dispatcher state x and the return of the probing mode

Hello, when trying to set a destination to inactive using dispatcher.set_state after setting the state to "i", when I do a "dispatcher.list" I see the state as "ix". The "x" state appears not to be documented. After a short time (seconds) the flags return to state "ip" (probing). According to

Re: [SR-Users] Kamailio and NAT

When you are running Kamailio behind a NAT you should use advertise parameter of listen address. According to documentation " A typical use case for advertise address is when running SIP server behind a NAT/Firewall, when the local IP address (to be used for bind) is different than the public IP

Re: [SR-Users] Kamailio and NAT

Hello, I finally were able to run my Kamailio behind NAT but in order to accomplish that I included: listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 52548 is the port my internet router change when doing NAT (5060->52548). I found this port sniffing traffic Conclusions at this point are:

Re: [SR-Users] Kamailio replies with SIP/2.0 500 Service Unavailable Under load

Hello, as an extra hint on top of Carsten's remarks: search your configuration for sl_send_reply("500", "Service Unavailable") or t_reply/send_reply with same parameters. Then you can identify better why that reply is sent. As Carsten pointed, such reply code and reason text is not coming from

Re: [SR-Users] Kamailio and freeswitch integration for SBC

What application is sending the 404? Cheers, Daniel On 13/01/16 21:43, malik sherif wrote: > > Any hint as to how to correct this issue? > > > 1 404 /UNALLOCATED_NUMBER/ Unallocated (unassigned) number [Q.850 > value 1] This cause indicates that the called party cannot be reached > because,

Re: [SR-Users] using re.subst with an $avp or $var

Hello, if you work with large data records that you want to get via kamcmd, then you have to increase the buffer sizes of ctl module -- see the readme of that module to identify the parameters to adjust. Cheers, Daniel On 11/01/16 20:22, Vik Killa wrote: > It seems dialplan module does not work

Re: [SR-Users] Kamailio and freeswitch integration for SBC

Kamailio is not generating another reply if freeswitch is sending one (unless enforced in config file). Are you sure the 404 is sent by Kamailio? What do you mean by "freeswitch is generated UN-allocated number"? Isn't free switch sending a sip reply in this case? Cheers, Daniel On 13/01/16

Re: [SR-Users] Kamailio and freeswitch integration for SBC

Any hint as to how to correct this issue? 1 404 UNALLOCATED_NUMBER Unallocated (unassigned) number [Q.850 value 1] This cause indicates that the called party cannot be reached because, although the called party number is in a valid format, it is not currently allocated (assigned).

Re: [SR-Users] dispatcher state x and the return of the probing mode

Hello, to complete: if you want destination to not to be pinged anymore and not used for routing, you have to disable (deactivate) it, not to set it in state inactive. There are three states: - active - inactive (at any time can become active based on ping activity) - disabled (it is going to be

Re: [SR-Users] Kamailio and freeswitch integration for SBC

Kamailio SIP is sending 404 not found since freeswitch is generated UN-allocated number, the call got rejected and goes to voicemail. Thank you again for your help Abdul From: Daniel-Constantin Mierla Sent: Wednesday, January 13, 2016 9:06

Re: [SR-Users] using re.subst with an $avp or $var

Hello, realoading is done in a separate structure, which is then swapped with the old one and the old one is freed -- like you said, a change of the pointer to the new structure. Cheers, Daniel On 11/01/16 19:49, Vik Killa wrote: > Hi Daniel, > We are experimenting with the dialplan module. > I

Re: [SR-Users] Kamailio and freeswitch integration for SBC

I am not familiar with your freeswitch config and what freeswitch should do. Also, I don't deal much with freeswitch in order to assist you with it, maybe other people here can help, if not, you can eventually ask on freeswitch mailing list. Cheers, Daniel On 13/01/16 18:15, malik sherif wrote:

Re: [SR-Users] Kamailio and freeswitch integration for SBC

is there a new to edit vars.xml file? I haven't touched this file but one of the warning about default password 016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. From: sr-users

Re: [SR-Users] Parser error when enabling siptrace with HEP functionality

Hi Sergey, thanks for testing and the feedback. The patch is also useful to have around for those than want to get the feature in 4.3 by one step, not cherry picking each commit. One more thing: are the local message properly captured? I mean the replies generated with sl_send_reply()/t_reply()

Re: [SR-Users] Kamailio and freeswitch integration for SBC

Perhaps this is a security message embedded in freeswitch code to instruct the admin to change default values coming in configs, otherwise you expose yourself to being hacked straight away. So open that file and change the default password. I have no clue about what is it, I am just interpreting

Re: [SR-Users] Core Dump during shutdown on SPARC64

Hi Daniel, See below: (gdb) p group->var_offset $1 = 64 (gdb) p mapping[i].offset $2 = 56 Thanks! Spencer > On Jan 13, 2016, at 10:18 PM, Daniel-Constantin Mierla > wrote: > > Hello, > > can you get the values for group->var_offset and mapping[i].offset in > frame 0? >

Re: [SR-Users] Core Dump during shutdown on SPARC64

Hello, can you get the values for group->var_offset and mapping[i].offset in frame 0? Cheers, Daniel On 14/01/16 05:41, Spencer Thomason wrote: > Hello, > I’m trying to get Kamailio running on Solaris 11 SPARC64 and I’m receiving a > bus error on shutdown. If needed, we can make SPARC

Re: [SR-Users] Core Dump during shutdown on SPARC64

Hello, hmm, they are multiple of 8, so they should be aligned to 64bits. Or maybe the 'block' variable value is not aligned to 8bytes ... Have you done any sip traffic via this kamailio instance. Is all ok at runtime? Cheers, Daniel On 14/01/16 08:01, Spencer Thomason wrote: > Hi Daniel, >

Re: [SR-Users] Kamailio and NAT

Hello, it looks like you have a symmetric nat router, so the allocated port is randomly selected. If you don't control the nat router to set a static forwarding rule or it doesn't provide the option to set static forwarding, then you are pretty much left with sniffing the traffic to discover the

[SR-Users] Core Dump during shutdown on SPARC64

Hello, I’m trying to get Kamailio running on Solaris 11 SPARC64 and I’m receiving a bus error on shutdown. If needed, we can make SPARC hardware available for testing. Thanks, Spencer Core was generated by `/opt/kamailio/sbin/kamailio -f /opt/kamailio/etc/kamailio/kamailio.cfg -P /syst'.

Re: [SR-Users] dispatcher state x and the return of the probing mode

hi. Sven. 1. Have you set modparam("dispatcher", "ds_probing_mode", 1)? If so, it will be moved to PROBING mode after ds_ping_interval 2. 'X' means "not any set". just a cross ;) 2016-01-13 16:17 GMT+03:00 Sven Neuhaus : > Hello, > > when trying to set a destination to

Re: [SR-Users] RTP Engine multiplexing and De-multiplexing

On 01/13/2016 04:26 AM, riko nir wrote: Hello, A call from a remote webrtc client is coming to (opensips+rtpengine). The media streams from the webrtc client is multiplexed. Can I use rtpengine to demultiplex the multiplexed streams and send it to other end as de-multiplexed SRTP traffic . This

Re: [SR-Users] Kamailio and freeswitch integration for SBC

Thanks again Daniel for replying. Now the call is failing with 404 not found and goes to voicemail. I was calling between 7632689991 and 7632689993, I looked the extensions on freeswitch, and look OK but it is possible I might have missed something. Freeswitch issues the following errors.