sr-users September 2007

sr-users@lists.kamailio.org

158 participants
248 discussions

Re: [OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by Iñaki Baz Castillo 11 Sep '07

11 Sep '07

First of all sorry for this post not being in the thread it belongs to (I've not the initial mail). > But, I don't want to use the Asterisk dtmf features, > I want to use the SIP INVITE/REFER features from the > UAC (actually polycom and snom phones). > > Can someboy shed me some light on this? You receive a call from Asterisk UAC and want to transfer (blind or attender) to other UACregistered in OpenSer. Of course you don't need Asterisk native transfer, SIP allows blind and attended transfer. The only you need is that your SIP devices allow both of them (unfortunatelly many SIP devices just allow blind transfer, but in most cases reading the device manual you'll find how to do a attended transfer). The point here is that Asterisk allows receiving a REFER, so don't worry for that (I use your same escenary with no problem and SIP transfers). Maybe you find useful this links: http://tech-invite.com/Ti-sip-service-4.html http://tech-invite.com/Ti-sip-service-5.html http://dev.openwengo.com/trac/openwengo/trac.cgi/wiki/SipTransfer Regards. -- Iñaki Baz Castillo

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Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 11 Sep '07

11 Sep '07

Hi, use ngrep or tcpdump to get a network trace from the proxy server, on the loopback interface. Regards, Bogdan Arya wrote: > Can you tell me how I can trace it? could this be an issue with my > router? its the Linksys Wr54G and its set on DMZ to the server > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Arya, > > I suspect you have a routing problem that cause the request to loop on > openser (via loopback interface) - openser keeps sending the > request to > itself. That's why you get a "Message to big" or "too many hops".... > > run a trace on lo to see if this is what's happening. > > regards, > bogdan > > Arya wrote: > > Hello everyone > > > > I have a problem connecting to OpenSER from my public IP. > > > > I can connect and make calls to OpenSER inside my LAN but when I > > connect from outside LAN I get the 513 Massage too big massage. > I than > > changed the openSER default config's section that checks a message > > size to a bigger number and I got the I get the "483 - too many > Hops" > > error. > > > > I changed the config back to default and tested it with ngrep and I > > tried connecting to the server from a server in another state. > > > > The SIP server is on 192.168.1.109 <http://192.168.1.109> > <http://192.168.1.109> and the > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> <http://192.168.1.100 <http://192.168.1.100>> > > > > And ngrep gave me these massages: > > > > > > > > > -- > Thank You

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[OpenSER-Users] Openser and Voice master
by Dan 11 Sep '07

11 Sep '07

Hi, Has anybody able to successfully able to route calls to voicemaster from openser. I tried all method but none worked. From username/password to IP based authentication to prefix based authentication. Regards Dan

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[OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by perl＠libkvm.org 11 Sep '07

11 Sep '07

Hello, I am configuring a platform unig OpenSER 1.2 with Asterisk 1.2. I have all the SIP phones registered on OpenSER. I use the realtime setup on Asterisk using a view on MySQL. I need to be able to process the attended transfer between two UAC registered on OpenSER, having the first received a call from the Asterisk side, so that the second UAC gets the call transferred. I also need to do this as a blind transfer. But, I don't want to use the Asterisk dtmf features, I want to use the SIP INVITE/REFER features from the UAC (actually polycom and snom phones). Can someboy shed me some light on this? I need to evaluate if this is possible, or if it needs lots of sources modifications. Thank you very much.

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[Serusers] ERROR: warning_builder: buffer size exceeded
by Kostas Marneris 10 Sep '07

10 Sep '07

Hello all, I have occasionally the following warning mesgs at my SER log : Sep 10 14:34:27 serhost ser[17957]: ERROR: warning_builder: buffer size exceeded Sep 10 14:34:27 serhost ser[17957]: WARNING: warning skipped -- too big I noticed that the R-URI in that case is : R-URI=<sip:XXXXXXXXXX@mydomain.com;user=phone;transport=UDP> when usually the R-URI is : <sip:XXXXXXXXXX@mydomain.com;transport=UDP> <sip:XXXXXXXXXXXXX@mydomain.com> Anyone seen those mesgs before ? Can this explained by the extra field 'user=phone' which the SIP UA adds at the R-URI of the INVITE mesg ? thanks Kostas

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[OpenSER-Users] Problem with rtpproxy
by Jeferson Prevedello 10 Sep '07

10 Sep '07

Hi Guys, When I originate a call for an user behind of a NAT the call is without audio in the two ways. I believe that the problem is related to rtpproxy, however did not find nothing of wrong in my configuration. Some tip? openser.cfg: # ------------- !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # Nathelper with RTPproxy service run: ps -aux | grep rtp root 26098 0.0 0.0 2532 428 ? Ss 10:40 0:00 /usr/bin/rtpproxy root 11143 0.0 0.0 2536 352 ? Ss 11:30 0:00 /usr/bin/rtpproxy -l xxx.xxx.xxx.xxx -s udp:xxx.xxx.xxx.xxx 7890 logs: Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11859]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11860]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11861]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11862]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Regards Jeferson

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[Serusers] Need SIP proxy to translate UDP<->TCP
by Thermal Wetland 10 Sep '07

10 Sep '07

Hello, I would like to hire someone to setup a SIP UDP<->TCP proxy using SER. SER will sit between an Asterisk box and Microsoft Exchange 2007. Asterisk speaks UDP & the Exchange box speaks TCP. I will make a fresh fully patched CentOS 5 box available for the work. Payment could be through PayPal or Rent a Coder or similar. I would expect an SER expert to be able to do this in less than 2 hours. Please let me know if you are interested - thermalwetland at gmail dot com. Thanks, Thermal

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Re: [OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Hi Bogdan I got it fixed. The problem is on this line: if(!t_relay("0x05")) if I change to if(!t_relay()) everything is ok. Don't know what is 0x05 mean. Regards, Halomoan On 9/10/07, Halomoan Chow <halomoan(a)gmail.com> wrote: > > Hi > > My Openser version is openser-1.2.0-notls and the Mediaproxy is > mediaproxy-1.9.0 > I was using the openser.cfg that was generated from > http://www.sipwise.com/wizard/ for Openser + Mediaproxy only. > > Just only simple call from A to B cause segmentation fault. > > I don't know how to locate the core file and get the backtrace :( > > Regards, > > Halomoan > > On 9/10/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > > > Hi Halomoan, > > > > provide full details about your openser version - version, how did you > > get it (source, binaries, etc). > > also locate the core file and get the backtrace. > > > > Regards, > > Bogdan > > > > Halomoan Chow wrote: > > > Dear All > > > > > > I'm running into segmentation fault error on Openser 1.2.x with > > > Mediaproxy. > > > What is the first thing I should take a look into to find out which > > > line or which module causing this segmentation fault? > > > Thank you. > > > > > > Regards, > > > > > > Halomoan > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

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Re: [Serusers] [Serdev] Release of SER 2.0 (and future 2.1)
by Greger V. Teigre 10 Sep '07

10 Sep '07

Ok, I conclude then that final release of SER 2.0 is aimed at October 15. That should allow us to align SER, serctl, SERWeb, rtpproxy, and SEMS. SER, SERWeb, and SEMS will have independent releases, and they will be aligned to be interoperable. SER, rtpproxy, and serctl will have a common release. The release will include the new buildsystem that was tested in release candidate 1. The config file and features available work with sercmd, serctl, as well as with SERWeb. SEMS support is planned, but pending some patches. All components will be combined in a one-file download with sources for all components, the SIP Express Bundle. I urge everybody with patches, bug reports, or tasks related to this release to prioritize those now! g-) Greger V. Teigre wrote: > Guys, > My last post concerned documentation. I suggested decoupling > documentation from the CVS to allow us to work independently with code > and docs. > > I suggest we move forward on the decoupling and set a date for SER 2.0 > release. My suggestion is that we target October 15. This will allow > people to allocate time to do the last clean-ups, bug-fixes, and patches. > > I also suggest that we set a release date for 2.1. AFAIU, there is > already quite a lot of code that has gone into the trunk and as 2.1 is a > minor release and we strive for ser.cfg compatibility with 2.0, we > shouldn't really wait too long. > > I suggest mid-January 2008. How does that work for you? > g-) > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > > >

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[OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Dear All I'm running into segmentation fault error on Openser 1.2.x with Mediaproxy. What is the first thing I should take a look into to find out which line or which module causing this segmentation fault? Thank you. Regards, Halomoan

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sr-users September 2007