sr-users June 2008

sr-users@lists.kamailio.org

124 participants
230 discussions

[OpenSER-Users] Playing the recorded RTP Session

by Chaitanya V.N.S

Hi, I added NATHelper module to openser and used RTP proxy I am also able to record the RTP session (Conersation between tow calls) using start_recording(); The rtp proxy is recording the conversation in file with rtp extenssion. The problem is with palying that rtp file. Can any one say me how to play that file. Thanks in advance, Chiatu.

16 years, 5 months

[OpenSER-Users] session timer behavior (sst)

by William Quan

Hi all, I have a simple (SIP Session Timer) sst call where 'Session-Expires' header (SE ) is present in the INVITE but not in the 200 OK constructed at the UAC. This indicates UAS does not support sst. 1. According to the spec (RFC4028), the stateful proxy remembers that SE was in the request, and adds the SE header to the 200OK, but fails to add the Require header (its not initially present either). Is there a way to modify the config to add a 'Require' header to a response when the session expires header was inserted by sst ? 2. Similarly in the mid-call 'refresh' txn, the addition of the SE header to the response does not work for me (v1.3.2). Shouldn't it automatically insert the SE header into the 200, like it did during call setup? When the SE is absent in the response the sst feature is thereafter disabled right? http://tools.ietf.org/html/rfc4028#section-8.2 -will

16 years, 5 months

[OpenSER-Users] peering module

by Klaus Darilion

Hi! I wonder if the new peering module is using a certain standard (which one?) or is a proprietary format used between openser and the radius server? regards klaus

16 years, 5 months

[OpenSER-Users] [OT] DMTF question (rfc2833)

by Lucio Maciel

Hi, This is offtopic, but maybe someone could help me. I have a problem with a costumer sending the DTMF via RTPEvent. They are sending just one packet per event, with the END bit set and the duration of the event, but some carriers are no recognizing this Event, only if you keep sending rtp events with the duration and then send a final event with the END bit set. This is valid, to send just one event with duration and END bit??? No. Time Source Destination Protocol Info 3 0.026233 xxx.xx.xxx.xx yyy.y.yyy.y RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) Frame 3 (62 bytes on wire, 62 bytes captured) Internet Protocol, Src: xxx.xx.xxx.xx (xxx.xx.xxx.xx), Dst: yyy.y.yyy.y (yyy.y.yyy.y) User Datagram Protocol, Src Port: pxc-spvr (4006), Dst Port: 13018 (13018) Real-Time Transport Protocol 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: Unknown (101) Sequence number: 51289 Timestamp: 1479952026 Synchronization Source identifier: 0x43256a7a (1126525562) RFC 2833 RTP Event Event ID: DTMF Two 2 (2) 1... .... = End of Event: True .0.. .... = Reserved: False ..00 0000 = Volume: 0 Event Duration: 1200 Thanks. -- Lucio Maciel <lucio(a)tesatelecom.com>

16 years, 5 months

[OpenSER-Users] About Callback Option

by Ali Jawad

Hi All Dear All is there a way to implement callback features in Openser ? Thanks

16 years, 5 months

[OpenSER-Users] Doubt about REGISTER by Third-party with OpenSER

by Marcio

Hi all again....someone could explain me? I have a doubt, about to constructing a REGISTER Request by third-party, if is possible with OpenSER`s modules??? and how? If can force a "Proxy A" to forward "Register" of the UAc to Proxy/Reg"B" on behalf of an UAc to Proxy/Registrar "B" directly... UAc(B) -----> Proxy/Registrar(A) ------------->Proxy/Registrar(B) 10.2 item rfc3261....."Registration on behalf of a particular address-of-record can be performed by a suitably authorized * third party*."... best regards, Marcio

16 years, 5 months

[OpenSER-Users] rtpproxy with openser and asterisk

by Martin-Gruber.External＠infineon.com

hi guys, I have the following scenario. I want to use Asterisk as voicemailserver, but need an rtp-Proxy because the my phone speaks srtp. I get SIP-Message from an PBX to openser and RTP Proxy. My question now is, how I can handle this case or how I can forward from rtpproxy to asterisk? just with rewritehostport? thanks in advance! regards Martin

16 years, 5 months

[OpenSER-Users] credentials not filled properly

by David Villasmil

anyone knows what this means? ERROR:auth:pre_auth: received credentials are not filled properly thansk!

16 years, 5 months

[Serusers] Gentilini, Paul is out of the office.

by PGentilini＠intellicare.com

I will be out of the office starting 06/17/2008 and will not return until 06/20/2008. I will have very limited access to mail, so please contact the CSC at 888-899-4227 if you need immediate assistance.

16 years, 5 months

[OpenSER-Users] Openser crash if using invalid flag in t_relay

by Ruchir

I had put t_relay("0x05") in relay route. When I test calling, openser dies with seg. fault at the time of t_relay. I found that allowed values are 0x01, 0x02 & 0x04 only but openser should generate error while starting in this case.

16 years, 5 months

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sr-users June 2008