sr-users April 2009

sr-users@lists.kamailio.org

83 participants
109 discussions

[Kamailio-Users] bidirectionnal chating and presence between sip and MSN

by rachid

Hello all , I want to have a bidirectionnal chating and presence between sip and MSN ,but i dont know how to configure kamailio thank you for your help -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

15 years, 2 months

Re: [Serusers] SER Support for SIP Diversion header

by Stefan Sayer

Hello, o Randy Jackson [04/22/09 19:49]: > Stefan, > > > > For a SER/SEMS configuration using the voicemail app, 4605 calls 4610. > 4610 doesn’t answer and call is forwarded to 9200. If the forwarded > call is routed to SER/SEMS (i.e. see example below) when the SER > (ser-2.0.0-rc1) receives the below INVITE which contains the Diversion > header in the message header portion of the request, what module and > methods do you use to retrieve the parameters from the Diversion header, > like the URI? 9200 does not have a vm account. The vm app will not > answer the call for account 4610 based on the below INVITE. Thanks in > advance. The question is: where do you get the email address from? If it's in DB I think you could get the user from the diversion header into avp with some clever textops and then do the DB query with it to get email address and user into P-App-Param header. If you are only using mode=box (only voicebox, no mail) though, it may be simpler to get the user in SEMS like with this patch to voicemail app: Index: AnswerMachine.cpp =================================================================== --- AnswerMachine.cpp (revision 1354) +++ AnswerMachine.cpp (working copy) @@ -531,10 +531,16 @@ throw AmSession::Exception(500, "voicemail: no email address"); } - user = get_header_keyvalue(iptel_app_param,"usr", "User"); + // user = get_header_keyvalue(iptel_app_param,"usr", "User"); + AmUriParser div_parser; + if (!div_parser.parse_contact(getHeader(req.hdrs, "Diversion"), 0, end)) { + ERROR("could not parse Diversion header"); + throw AmSession::Exception(500, "voicemail: no user found"); + } + user = div_parser.uri_user; if (!user.length()) user = req.user; - + sender = get_header_keyvalue(iptel_app_param, "snd", "Sender"); if (!sender.length()) sender = req.from; btw, isn't Diversion supersedet by History-Info? BR Stefan > > > > > > INVITE sip:9200@192.168.1.1;user=phone > <mailto:9200@192.168.1.1;user=phone> SIP/2.0 > > Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKac816850588;alias > > Max-Forwards: 70 > > From: <sip:4605@192.168.1.2>;tag=1c816838653 > > To: <sip:9200@_192.168.1.1_;user=phone> > > Call-ID: 816837947271200020365(a)192.168.1.2 > > CSeq: 1 INVITE > > Contact: <sip:4605@192.168.1.2;transport=tcp> > > Supported: em,100rel,timer,replaces,path,resource-priority > > Allow: > REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE > > *Diversion: <sip:4610@192.168.1.1>;reason=no-answer;screen=no;privacy=off* > > User-Agent: > > Content-Type: application/sdp > > Content-Length: 270 > > > -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

15 years, 2 months

[Kamailio-Users] SR-Users-ES maillist

by Iñaki Baz Castillo

Hi, for the spanish speakers community there is a maillist about Sip-Router: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es This maillist has been merged with the previous Kamailio Users-es maillist, so the maillist history remains. Best regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

15 years, 2 months

[Kamailio-Users] New listen IP without restart

by catalina oancea

Hello everybody, Does anybody know if there is a way to add a new listen line in kamailio.cfg and make kamailio start listening on that IP without restarting the server? Thanks a lot Catalina

15 years, 2 months

[Serusers] is serusers@lists.iptel.org mailing list still in use?

by Leon Li

Hi, I am just wondering whether serusers(a)lists.iptel.org is still the one to request support? I didn't received any emails in last several days. Cheers Leon

15 years, 2 months

Re: [Kamailio-Users] Multiple SIP Proxy Environment / Socket Information

by Iñaki Baz Castillo

El Martes, 21 de Abril de 2009, Brandon Armstead escribió: > Inaki, Please, mantain the thread in the maillist. > Thanks for the input -- that does look similar to what I'm looking to > do, however not 100%. From looking at the RFC it looks as if there is > always a certain path being traversed, however in my case, its more of a > "this is not the server that holds this user location socket binding, lets > forward to the server that does". > > So in essence: > > UA1 can register to P1, P2, P3, and so on..... > > However it does not register to P3 via the means of UA1 REGISTER -> P1 -> > P2 -> P3 (REGISTRATION). > > It simply registers directly to P1, or P2, or P3, etc.... > > In which case upon user lookup: lookup("location"), I want to be able to > say "Hey I do not hold the socket for this registration, forward to server > that does to do the actual lookup(location) and t_relay() to the UA. Yes, it's a different scenario. You could do some trick: - When P1 receives a REGISTER it sets a bflag for it (bflag 1). - When P2 receives a REGISTER it sets a bflag for it (bflag 2). - When P3receives a REGISTER it sets a bflag for it (bflag 3). When P1 receives a request and performs location for that AoR, it will also extract the bflags, so P1 checks if bflag 1 is on. If not and bflag 2 is on, then it routes (by setting $du => without changing the resolved RURI) the request to P2. :) -- Iñaki Baz Castillo <ibc(a)aliax.net>

15 years, 2 months

[Kamailio-Users] Disable auto ACK

by kévin l'heveder

Hi, i want to generate my own ACK after a OK ( After ringing). but even if i drop the OK in my reply_route a ACK is auto generated by Kamailio and sent How could i prevent this behavior ?? I found some clues in the sources : *#define no_autoack(_t_) ((_t_)->flags&T_NO_AUTOACK_FLAG)* But i didn't see how i could flag a message to disable AUTOACK If someone have an idea or THE response, that would be great!! Thanks. Koon

15 years, 2 months

[Kamailio-Users] configuration

by shiva prasad

Hi All, I've successfully installed kamailio 1.5x version on fedora 9 os, could you please let me know is it possible to configure Ekiga(client) to connect kamailio server,. if possible, could you please let me know how to configure the client? Thanks in advance. -- Regards, Shivaprasad BS

15 years, 2 months

[Kamailio-Users] help using rewrite prefix

by Carlos A. Alvarez

I need help setting up carrierroute using the rewrite prefix and the strip function. It seem like every time I use the Rewrite Prfix function the carrierroute module will rewrite the prefix, but then I don't get certain reply messages on kamailio. At least from what I can see on the log. INFO:carrierroute:cr_do_route: uri 5555550001 was rewritten to sip:0115555550001@aslab.commx.net, carrier 1, domain 10 INFO:carrierroute:rewrite_uri_recursor: URI or prefix tree nodes empty, empty rule list INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri 0115555550001, carrier 1, domain 10 Is this normal behavior? I also noticed that some messages coming back from the host are not being processed by kamailio. For example on this particular call I purposely get a 604 from the server. The thing is I can't process a failure route for this message, because kamailio doesn't see is, even though it is being sent back to the server. Also when I use the Strip function is looks like carrierroute does a second lookup on the route table. In the example below I am calling a number using the prefix 551... The route table is below. +----+---------+--------+--------------+-------+------+------+-------+-----------------+----------------+----------------+------------------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+--------------+-------+------+------+-------+-----------------+----------------+----------------+------------------------+ | 30 | 1 | 1 | 551 | 0 | 0 | 1 | 2 | 10.10.10.1 | | | LD-1 | 31 | 1 | 1 | 1 | 0 | 0 | 1 | 0 | 10.10.10.1 | | | LD | Log for that call INFO:carrierroute:cr_do_route: uri 5515555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 INFO:carrierroute:cr_do_route: uri 15555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 Carrier found!! ("1", "131.226.0.144", "15555550004", "15555550004", "COMMX-LD"); Now please note that if I remove route id 31 then the call log looks like it is below: INFO:carrierroute:cr_do_route: uri 5515555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 INFO:carrierroute:rewrite_uri_recursor: URI or prefix tree nodes empty, empty rule list INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri 15555550004, carrier 1, domain 1 Please help!!! Carlos It will also help if there was complete routing example using carrierroute v 1.5.0, since the one in the wiki is outdated. Thanks.

15 years, 2 months

Re: [Kamailio-Users] bidirectionnal chat and presence sharing between a SIP and a jabber user

by Daniel-Constantin Mierla

Please cc all the time to mailing lists. Emailing privately will make the message dropped. Also, mailing list has the benefit of providing access to other people that might know better the answer. Thanks, Daniel On 04/20/2009 01:18 PM, rachid wrote: > *Hello, > I use the purplemap table to mapping sip uri and jabber client,so i > want to have a bidirectionnal chat and sharing between a dip and > jabber users > > I tried with purple modules* > > 2009/4/20 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 04/17/2009 11:22 AM, rachid wrote: > > Hello all, > > someone can help me to configure kamailio to i was able to have a > bidirectionnal chat and presence sharing between a SIP and a > jabber user ? > thank you > > what is the problem? Did you get or not any communication between > sip and xmpp? Have you tried with xmpp or purple modules? > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > > > -- > ------------------------------------------------------------ > BAHA Rachid > Elève ingénieur en 3ème année à l'INPT, Rabat > GSM:(+33) 0619787609 > > -- Daniel-Constantin Mierla http://www.asipto.com/

15 years, 2 months

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sr-users April 2009