sr-users February 2006

sr-users@lists.kamailio.org

189 participants
374 discussions

[Serusers] Users in ENUM
by Anderson Alves Albuquerque 21 Feb '06

21 Feb '06

If I have a prefix 1100 and I send call to 11000202, how can I use ENUM this? If I have the line below, I would need to put username but I don´t want this: $ORIGIN 0.2.0.0.0.1.1.e164.arpa. 2 IN NAPTR 1 10 "u" "E2u+sip" "!^.*$!sip:10.10.10.1<r(a)10.10.10.1>; sip:john@10.10.10.2!" . Suppose, I send to 11000202 or 11000203 I want send call to 10.10.10.2 but I want to use the e164 original. If have a lot of users I would need to put all the users in ENUM, but I don´t want this. How can I do this?

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[Serusers] Forwarding to port 5060 when UA behind NAT
by Arek Bekiersz 21 Feb '06

21 Feb '06

Dear list, I have situation where SIP UA is behind NAT, that maps everything to ports above number 61000. I use nathelper, rtpproxy and SER 0.8.12. Everything works fine. Until I have UA that does not send "Contact" field in BYE (or OK for BYE) during call termination. In this case SER is forwarding responses not to NAT-ed port, but to port 5060. QUESTION: What can be done to correct this situation? Is this improved in new SER releases? Sequence is as follows: IP_of_A:61001 -> IP_of_proxy:5060 BYE sip:B@IP_of_proxy loose_route ... IP_of_proxy:5060 -> IP_of_B:5060 BYE sip:B@IP_of_B ... IP_of_B:5060 -> IP_of_proxy:5060 SIP/2.0 200 OK ... IP_of_proxy:5060 -> IP_of_A:5060 <WRONG! Should be 61001> SIP/2.0 200 OK ... It does not happen when terminating UA sends "Contact" field in first BYE. -- Regards, Arek Bekiersz

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[Users] TLS stability
by Klaus Darilion 21 Feb '06

21 Feb '06

Hi! Recently on the serdev mailinglists there were considerations about the stability of the TLS code: http://mail.iptel.org/pipermail/serdev/2006-January/006608.html I know there was a fix to disable openssl compression. Was this the only known problem? Is TLS now running stable? Has someone ever used the TLS stack successfully under load? regards klaus

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Re: [Serusers] RV: serweb error
by Karel Kozlik 21 Feb '06

21 Feb '06

HI, set fifo_mode = 666 in your ser.cfg Karel Guty napsal(a): > > > Hello! > > > > I’m having a trouble with my serweb. > > > > I can access to the user interface login screen but when I log on I > receive the next error: > > > > *Warning*: fopen(/tmp/ser_fifo): failed to open stream: Permission > denied in */home/gustavo/ser/serweb-0.9.4/html/functions.php* on line *340* > > *Warning*: Cannot modify header information - headers already sent by > (output started at > /home/gustavo/ser/serweb-0.9.4/html/functions.php:340) in > */home/gustavo/ser/serweb-0.9.4/html/page.php* on line *35* > > * * > > Thanks!!!!!! > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] RV: serweb error
by Guty 21 Feb '06

21 Feb '06

Hello! I'm having a trouble with my serweb. I can access to the user interface login screen but when I log on I receive the next error: Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied in /home/gustavo/ser/serweb-0.9.4/html/functions.php on line 340 Warning: Cannot modify header information - headers already sent by (output started at /home/gustavo/ser/serweb-0.9.4/html/functions.php:340) in /home/gustavo/ser/serweb-0.9.4/html/page.php on line 35 Thanks!!!!!!

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[Users] caller number removal
by unplug 21 Feb '06

21 Feb '06

In the situation below, a call made from PSTN to openser. PSTN -----> openser ---------> UA1 How can I remove the caller such that UA1 can't show the caller number in her display?

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[Users] preparation of patch release 1.0.1
by Daniel-Constantin Mierla 21 Feb '06

21 Feb '06

Hello everybody, a new package building of CVS branch rel_1_0_0 (release v1.0.0) will take place soon. This comes as an update to release 1.0.0 which will include the fixings occurred since 1.0.0 was released. We like to collect all issues of v1.0.0 you are aware of, to be fixed before package generation. Next is a list of the issues that should be fixed: - TLS - when TLS connections is refused, immediate reply instead of waiting for timeout - TCP - when the call is initiated over TCP but the contact advertised is UDP, the BYE from callee is not routed properly (it is tried TCP due to double RR) - record_route_preset() parameter size - ruri matching of canceled transactions (see: http://openser.org/pipermail/users/2006-February/002951.html) Any feedback is appreciated. Cheers, Daniel

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RV: [Users] Multiple Interfaces rtpproxy
by Fernando Rodriguez 21 Feb '06

21 Feb '06

Could a solution be using multiple domains per interface of the machina so i can run diferent instantes of the media proxy ?? _____ De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre de Fernando Rodriguez Enviado el: Lunes, 20 de Febrero de 2006 11:32 a.m. Para: users(a)openser.org Asunto: [Users] Multiple Interfaces rtpproxy Hello, I have a running openser on a machine with multiple interfaces im using mediaproxy but I was only able to set it up using 2 interfaces .. How can I set it up to have more working interfaces 1 lan 1 dmz 1 wan Right now it only mediaproxies the wan and dmz interfaces But I want to be able to mediaproxy betrwen the lan and wan and dmz to lan Does any one has a set up like this and will be able to guide me on seting this up .. Thanks a lot for your help Fernando Rodriguez V. frod(a)aitelecom.net

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[Users] Parameter domain_prefix in AVPOPS
by Papadopoulos Georgios 21 Feb '06

21 Feb '06

Hello, Just as a sugestion, wouldn't it be useful to add a parameter like "domain_prefix" in AVPOPS? thank you George

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[Users] add/rewrite a tag/parameter in SDP
by Lenir 21 Feb '06

21 Feb '06

Hey guys, What is the best way to modify the SDP message to always include (if not replace) the a= tags in a SDP message? My problem is specifically with g729, according to the RFC, if the parameter: "a=fmtp:18 annexb=yes" is present or missing then it is assumed that g729b (g729 with VAD) and if "a=fmtp:18 annexb=no" is present then the devices will use g729a (g729 with VAD Disabled). My problem is that some cisco gateways and ATAs behave differently (of course Cisco always doing it their way), if "a=fmtp:18 annexb=no" is missing then it assumes that g729a will be used, instead of following the RFC where it should negotiate g729b. Also, asterisk doesn't support g729b so I want to avoid using g729b everywhere in my network. So what is the best way or is there a way, to either: a) insert "a=fmtp:18 annexb=no" if is not present b) replace "a=fmtp:18 annexb=yes" for "a=fmtp:18 annexb=no" if it is present Thanks in advance!! Lenir p.s. Please see the below excerpt from RFC 3555 regarding how to specify G729. RFC 3555 MIME Type Registration of RTP Payload Formats July 2003 4.1.9. Registration of MIME media type audio/G729 MIME media type name: audio MIME subtype name: G729 Required parameters: None Optional parameters: ptime, maxptime annexb: indicates that Annex B, voice activity detection, is used or preferred. Permissible values are "yes" and "no" (without the quotes); "yes" is implied if this parameter is omitted

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sr-users February 2006