Re: [SR-Users] RTPENGINE rtp errors

2016-06-20 Thread Dmitry
Actually the only difference that I see in RTP dumps - is that the Yate client SIP softphone sends the first RTP packet as : Version = 0; and other packets are normal(comply to the rfc3550. Probably this is the cause of the problems. On Tuesday, June 21, 2016 9:46 AM, Dmitry wrote:

[SR-Users] RTPENGINE rtp errors

2016-06-20 Thread Dmitry
Hi 1)I encountered such errors when I make an internal call between 2 yate client SIP softphones:rtpengine is used and no audio: | Jun 19 10:02:35 kazootest2 rtpengine[2066]: [918508371@kazootest2.siplabs.local port 30942] Error parsing RTP header: short packet (header) when I turn off the rtpe

Re: [SR-Users] Kamailio consult CNAM server through SIP Subscribe.

2016-06-20 Thread Daniel-Constantin Mierla
Can you try with master branch or backport the last two patches from tm module? I pushed two commits that should catch and handle better this case. Cheers, Daniel On 20/06/16 18:14, Daniel-Constantin Mierla wrote: > > Hello, > > it seems it tries to generate an outgoing cancel for the suspended

Re: [SR-Users] Kamailio consult CNAM server through SIP Subscribe.

2016-06-20 Thread Daniel-Constantin Mierla
Hello, it seems it tries to generate an outgoing cancel for the suspended branch. I will check the code, likely there has to be added condition for this cases. Is the 487 reply for invite sent back? Also, the 200ok for cancel? Cheers, Daniel On 20/06/16 16:38, José Seabra wrote: > Hello, > > I

Re: [SR-Users] Equivalent of OpenSIP's binary interface in Kamailio

2016-06-20 Thread SamyGo
Hi, What kind of design you've in mind when thinking about BIN interface ? What are the things that you want to replicate ? Just call dialog states ? Given some more usage clarification, someone might be able to guide further on what can be used. Regards, Sammy On Sat, Jun 18, 2016 at 2:15 AM,

Re: [SR-Users] Kamailio consult CNAM server through SIP Subscribe.

2016-06-20 Thread José Seabra
Hello, I'm attaching more logs to this email regarding to the issue on SIP CANCEL to an INVITE that is suspended. If do you think that i should open an issue on git regarding to this let me know. Thank you for your help. Best Regards José 2016-06-15 14:42 GMT+01:00 José Seabra : > Hi Daniel

Re: [SR-Users] No other running SIP apllication on the machine.//答复: Kamailio ERROR "Address already in use" in tcp_init().

2016-06-20 Thread Zaka
So it means you are already running one instance of kamailio. If this is the case then killall kamailio and try again. REGARDS, ZAKA On 20 Jun 2016 15:18, wrote: > Hello, > > I don’t find other running SIP application except kamailio on the machine, > and the following information appear: > > [

[SR-Users] No other running SIP apllication on the machine.//答复: Kamailio ERROR "Address already in use" in tcp_init().

2016-06-20 Thread joey . ren
Hello, I don’t find other running SIP application except kamailio on the machine, and the following information appear: [root@localhost kamailio]# netstat -anul and grep on 5060 Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address   Foreign Addre

Re: [SR-Users] Kamailio dispatcher module - keep alives

2016-06-20 Thread José Seabra
Hi Daniel, I have enabled kamailio debug=3 and i can see the next hop of sip OPTION before it enter in event_route[tm:local-request], in this event route i'm changing the $du variable on the script but OPTION still going to address configured on dispatcher list instead of the address set on $du, a

Re: [SR-Users] Compiling kamailio with custom openssl

2016-06-20 Thread Cibin Paul
Hello Daniel, Sorry for late response. I tried adding the following entry in modules/tls/Makefile ifneq ($(SSL_BUILDER),) DEFS += -I/usr/local/ssl/include LIBS += -L/usr/local/ssl/lib Starting kamailio, giving following error. I am using kamailio-4.4.1 with openssl 1.0.2h E