sr-users April 2008

sr-users@lists.kamailio.org

141 participants
188 discussions

[Serusers] SER and ptime ?
by Mike Trest - Personal 03 Apr '08

03 Apr '08

Hi, I have registering phones (Linksys) that use g711 or g729. I have a media question in a situation when I attempt to do pure-proxy without media gateway or proxy. Because these user are always messing with their adapter settings, I have variable call setup when I forward certain destinaitons because of ptime miss-match between Linksys and destination. Is it possible to accpet / reject / revise the inbound call when the "ptime" is not specifically 20ms or 30ms (or any specific value) before t_relay on to destination? I see the ptime value in SDP but before I embark on journey trying to "manage" call setup based on that, I want to know if others are working with this issue. ..mike..

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[OpenSER-Users] [Fwd: contact uri]
by Amit Vijayvargiya 03 Apr '08

03 Apr '08

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[Serusers] How to return a string value of an exported functions from a module to be used in ser.cfg?
by lin9k＠yahoo.com 03 Apr '08

03 Apr '08

Hi, I am thinking to write a module and export a function to be called from ser.cfg. I need the exported function to return a string value to be used by another function in ser.cfg. How do I do that in ser.cfg? For example: in ser.cfg: function1(my_function()); in my_module: str *my_function(){ ... return my_string; } Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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[OpenSER-Users] Res: My first OpenSER post
by Flavio Goncalves 02 Apr '08

02 Apr '08

Hi Yakout, Check SerMyAdmin at http://sourceforge.net/projects/sermyadmin. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Yakout Esmat <yesmat(a)xtra.co.nz> Para: users(a)lists.openser.org Enviadas: Terça-feira, 1 de Abril de 2008 16:08:44 Assunto: [OpenSER-Users] My first OpenSER post Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] rtpproxy address filling
by Andres 02 Apr '08

02 Apr '08

Hi, I have a question regarding the way rtpproxy handles 'address filling'. After a session has been created, the rtpproxy pre-fills the caller and callee's addresses and we see that in the rtpproxy output like: pre-filling caller's address with 192.116.246.234:41000 pre-filling callee's address with 192.116.246.234:20000 Then when it sees the actual packets coming in from a different source port, it updates the address and we see it in the log like: callee's address filled in: 192.116.246.234:1024 (RTP) The audio then flows fine both ways. My question is, what would happen it the actual packets came in from a different IP while the rtpproxy was waiting between the state of 'pre-filling' and 'address filled' states? Will the rtpproxy accept such a change that includes a new IP? Will it ignore packets from a different IP and continue the session normally? Or will it abort the session completely? Thanks, Andres http://www.telesip.net

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[Serusers] Fwd: Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Stefan Thanks a lot for your prompt response. I can hear the announcement now. Previously my audio setup was not correct. Best Regards, Frq On Wed, Apr 2, 2008 at 12:11 PM, Stefan Sayer <stefan.sayer(a)iptego.com> wrote: > Hello, > > frq ser wrote: > > > Hi > > > > Thanks for your reply Stefan and my apologies for coming back to this > > issue after such a long time. As you suggested, I included SDP headers in my > > Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the > > announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 > > OK message, but no audio was being played. In the log, I got these messages > > (just including the important messages)... > > > > .. > > (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: > > dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying > > '/usr/local/lib/sems/audio/192.168.3.208/200.wav < > > http://192.168.3.208/200.wav>' (17358) DEBUG: getAnnounceFile > > (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' > > .. > > .. > > (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global > > address: 192.168.3.208 <http://192.168.3.208/> (17358) DEBUG: > > negotiate (AmSession.cpp:262): remote party doesn't support telephone events > > > > Seeing the last line, I then used an Xlite softphone (instead of sipsak) > > to contact SEMS. Again, the call gets established for about 10 seconds and > > then just gets disconnected and no audio can be heard. Nothing in > > > To me the logs look normal. seems like it plays the announcement and then > hangs up, as it should. the default_en.wav (or 200.wav, if you've copied it > there), is only a few seconds long. are you sure that audio is setup > correctly (volume etc), i.e. does the xlite test call work? if you call e.g. > sip:conference@iptel.org <sip%3Aconference(a)iptel.org>, do you hear > something? > > can you run tcpdump -i any or run wireshark to see whether packets flow > between SEMS and xlite? > > you could also PM me a wireshark dump so i could have a look at what's > going on. > > Stefan > >

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Re: [Serusers] Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Thanks for your reply Stefan and my apologies for coming back to this issue after such a long time. As you suggested, I included SDP headers in my Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 OK message, but no audio was being played. In the log, I got these messages (just including the important messages)... .. (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying '/usr/local/lib/sems/audio/192.168.3.208/200.wav' (17358) DEBUG: getAnnounceFile (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' .. .. (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global address: 192.168.3.208 (17358) DEBUG: negotiate (AmSession.cpp:262): remote party doesn't support telephone events Seeing the last line, I then used an Xlite softphone (instead of sipsak) to contact SEMS. Again, the call gets established for about 10 seconds and then just gets disconnected and no audio can be heard. Nothing in the SEMS log (see below) suggests that there is any problem (as far as I can understand). I've checked that the file /usr/local/lib/sems/audio/200.wav is present and I can play it directly in Linux using play command. (The file 200.wav is just a copy of default_en.wav file) I just want to know why the announcement file is not being played and why the call between softphone and SEMS just gets disconnected after a while. I am sure it would be something very basic :( Thanks a lot for your help Frq

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[OpenSER-Users] openser social networking events - April-June 2008
by Daniel-Constantin Mierla 02 Apr '08

02 Apr '08

Hello, in the period of time April - June 2008, there will be a series of openser social networking events organized in different places world wide, during the time some other Open Source or VoIP events take place in the same location. The events are most probably in the form of a dinner, organized together with local representatives of openser community. The events are intended to be rather small size, to allow each one to know the others, in a nice and friendly discussion environment. The primary goal is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to any of these events is free, everybody pays for its expenses. Anyone interested in openser, sip or voip in general is welcome to join. Next events are: - Bucharest, Romania, April 12, 2008, tied up with ROSDEV2008 (http://www.rosdev.ro - Romanian site only) - Orlando, FL, USA, April 22, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.edvina.net) - Barcelona, Spain, May 6, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.avanzada7.com/) - Vienna, Austria, May 23, 2008, on the way to AsteriskTag (http://www.asterisk-tag.org/wiki/Hauptseite) - Berlin, Germany, May 28, 2008, tied up with LinuxTag (http://www.linuxtag.org) - Amsterdam, Netherlands, June 3, 2008, tied up with VoN.x Europe (http://www.von.com/web/index.php) The hour and place for each event will be announced few days before as we know the number of attending people and we make the reservation. Please send email to me to reserve a place, although you are welcome to come at any time and without prior notice, you may find that there is no more available seat to join the table. If you live in the surroundings of these cities, and want to help with organization, you know a nice and quite place suitable for the event, let me know. Cheers, Daniel -- http://www.asipto.com

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[OpenSER-Users] Building Carrierroute Tables
by Douglas Garstang 02 Apr '08

02 Apr '08

So, I was kind of wondering how those using carrierroute where populating their routes. I was hoping to build a cost table that contained the costs for all our routes and all our carriers, and then construct the carrierroute table as a view derived from the cost table. I guess my SQL (and those around me I asked about it) isn't that good, because I haven't been able to come up with a way to do it yet. Doug. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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[OpenSER-Users] My first OpenSER post
by Yakout Esmat 01 Apr '08

01 Apr '08

Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM

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sr-users April 2008