sr-users April 2007

sr-users@lists.kamailio.org

177 participants
272 discussions

[Users] Debian package of Openser 1.2.0 with TLS compiled in?

by Kerker Staffan

Hi are there any debian packages released with the TLS support compiled in? i found all other packages for debian, but none with TLS support. best regards /Staffan Kerker --- Staffan Kerker, Saab Communication Ljungadalsgatan 2, 35180 Växjö, Sweden p. +46 470 42185 c. +46 705 391365 m. staffan.kerker(a)saabgroup.com w. http://www.saabgroup.com

17 years, 7 months

[Users] One way audio problem (Urgent)

by kyneoh＠dallab.com

Hello all, Ive been having a problem i cant seem to figure out, and i can see from various Google and list searches that i havent been the only one having this, but havent been able to find a solution yet for more than 3 weeks. Ive a bit frustrate to solve current problem. Ive got 4 UAs here. User agent A & C & D are located under WLAN, and user agent B located in the same LAN as SIP server and STUN server. User agent A & C connected to a restricted NAT that used STUN server to discover its own public IP. User agent A & C are mobile phone. (OS : window mobile 5.0) User agent D is softphone install in laptop. (OS : window XP) When calls gets sent from Box A to Box C (remote) i get a problem with the one-way audio(incoming caller can hear my voice, but i cant hear them). While, calls from Box A to Box D was successful. Both calls are form the same structure, how come calls work on A to D ??? (D) | (A)-----[NAT]-----------[NAT]---------------[SIP Proxy]-----(B) | | (C) [STUN server] *** SIP server and STUN server have assigned different public IP adress. SIP server : OPENSER + RTP Proxy in OPENSUSE Linux. Box A, C : Ageet softphone. Box B ,D : X-Lite softphone. Hope some of you out there are able to help me out a bit, its getting a bit annoying not being able to hear people calling me. Regards, Kum

17 years, 7 months

[Users] Proxy server only

by Lucas Diaz

This is my scenario: I got Sip devices using as Registrar and Proxy a public service from Internet. I want to OpenSER in the middle so I can track and manage the calls. How can I configure OpenSER? Do I need to do something in the external Server? Any help would be apreciated. Lucas.

17 years, 7 months

[Users] memlog question

by Papadopoulos Georgios

Hello all, I have a question about what is the best way to enable logging in openser 1.1.x. I want also to be able to get the memory dump so I have compiled with the appropriate flags and in my config I have debug=2 memlog=1 log_stderror=no This outputs everything into /var/log/messages, but the problem is that the memory dump section is messed up (lines do not seem to be in the right order). Also it takes a lot of time to stop openser with this configuration and I end up killing it. Is this because dumping all the memory to syslog is very slow? Is there a better way to do this? thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

17 years, 7 months

[Users] users in the same OpenSer cant call each other

by Joao Pereira

Hello My Openser users can call PSTN numbers through a PSTN gateway. But they cant call each other dialing the usernames. Where is the bug in my config? here is my openser.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; # -------------------------------------------------- # Registration # -------------------------------------------------- if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myserver.com", "subscriber")) { www_challenge("myserver.com", "0"); exit; }; save("location"); exit; }; } if (is_method("INVITE")) { # Route E.164 numbers to PSTN Gateway if (uri=~"sip:\+[1-9][0-9]*@myserver.com" ) { if (isflagset(29)){ route(2); return; } } # Route E.164 numbers to PSTN Gateway if (uri=~"sip:[1-9][0-9]*@myserver.com" ) { if (isflagset(29)){ route(2); return; } } } # method = invite route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } # Gateway PSTN route[2] { rewritehost("gateways_IP"); t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } Thanks for the help Joao Pereira

17 years, 7 months

[Users] help plz

by shruu thii

hi all, i'm using fc4.i've installed openser 1.0.1.how should i enable this for ipv6. i'm using linphone client.how do i register? i wanna do pc-pc calling on lan. thankyou. shruthi --------------------------------- Heres a new way to find what you're looking for - Yahoo! Answers

17 years, 7 months

[Users] force_rtp_proxy() and t_write_unix()

by Victor Gamov

hi! Can anybody explain me: when I do force_rtp_proxy("c") and then t_write_unix() which request body write to socket -- original or rewrited by force_rtp_proxy() ? Now I trying to configure OpenSER-1.2.0 to work with SEMS and I found that SEMS get original SDP with internal IPs. Is it my OpenSER misconfiguration? Thanks! -- CU, Victor Gamov

17 years, 7 months

Re: [Serusers] re: Malformed CONTACT field in 302 response

by Olaf Bergmann

Jignesh Gandhi wrote: > Thank you for your reply. > > Options #1 and 2 are not possible. > I am looking at how to do Option 3. Can you tell me as to > how to go about setting the q value ? Sorry, I am not sure how to do this in SER 0.8.x. Maybe someone else out there? > thanks, > Jignesh gandhi > > > On 4/10/07, *Olaf Bergmann* <Olaf.Bergmann(a)freenet-ag.de > <mailto:Olaf.Bergmann@freenet-ag.de>> wrote: > > Jignesh Gandhi wrote: > > > I recently posted almost the same question. > > And you got a number of good suggestions how to get around your > problem. Largely, the options were: > > - Make your softswitch vendor fix the bug. > - Use another softswitch. > - Adapt your custom function gl_redirect() to create a Contact field > with angle brackets. > - Set the q value. > > Which one did you check out? Did it work? Why not? What about the > other options? > > Regards, > Olaf > > > > > -- > Jignesh Gandhi > jigpgandhi(a)gmail.com <mailto:jigpgandhi@gmail.com> Regards, Olaf

17 years, 7 months

[Users] One way audio problem (Urgent)

by kyneoh＠dallab.com

17 years, 7 months

[Users] Terminate Call

by Javier

How to I can teminate call when prepayed debit run out? __________________________________________________ Preguntá. Respondé. Descubrí. Todo lo que querías saber, y lo que ni imaginabas, está en Yahoo! Respuestas (Beta). ¡Probalo ya! http://www.yahoo.com.ar/respuestas

17 years, 7 months

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sr-users April 2007