sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

RE: [Serusers] problem establishing conversation between two sip users

by Klaus Darilion

you can find some hints on: http://iptel.org/ser/problems/ the first thing you should do is to watch your network traffic (e.g. using ethereal) to find out who sends messages to whom. Furthermore, xlite has a debug window which shows all send and received messages. (press F9). regards, Klaus > -----Original Message----- > From: Ziying Sherwin [mailto:sherwin@lhc.nlm.nih.gov] > Sent: Friday, January 23, 2004 4:55 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] problem establishing conversation between > two sip users > > > > > We are now running ser 0.8.12 daemon on Solaris 2.8 platform > without persistency > support. When we do the testing, we tried to establish a > conversation between > two X-lite 2.0 clients built on Mac OS X. After we started > the X-lite, we > configured it to use our sip server. The user can log into the sip > server properly. However, when we tried to initiate a > conversation from one > X-lite to another, even though both of them are logged into > the same host, > they could not reach each other. The invitaion finally timed out. > > We also tried to initiate a conversation from X-lite to > Windows messenger. It > works fine, but we could not initiate a conversation from > Windows Messenger to > X-lite. > > Do anyone have similiar experience? Is there anyway to check > sip users' > activities on the ser side? How do we know whether there is > communication > between two users? We tried to use "serctl moni" to monitor > the activity, > but the log is long and confusing. Is there any > documentatiion that we can > refer to? > > Thanks in advance. > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

20 years, 10 months

[Serusers] Don't Want To Be Busted ? ? ? . . . . . . . . tutored

by Sonia.g

20 years, 10 months

RE: [Serusers] SIP server compatibility

by Klaus Darilion

Hi Annie! I read that you're using Windows CE. I haven't knowed that also the CE platform supports SIP. Is it the same RTC-Api which is included in Windows XP. Regarding your problem - try to capture the network traffic, if there are SIP messages sent between your client and the proxy. If yes, attach them to your email. regards, Klaus > -----Original Message----- > From: Annie Sasidar [mailto:asasidar@mail.unomaha.edu] > Sent: Thursday, January 22, 2004 7:41 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] SIP server compatibility > > > > > > > Hi, > I am working on windowsce.net4.2. The voip architecture in this OS is > based on SIP (RFC2543). I have been trying to register on the > Linux SIP > server running ser using a Provisioning.xml file which has the details > required. I am getting the error " Unable to register, check server > settings". I am running the ser-0.8.12 version on the linux > box which is > based on RFC 3261. Do you think there is a compatibility > issue here? Does > ser allow clients which are based on older RFC's to register? > Any pointers will be greatly appreciated. > > Thanks, > Annie > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

20 years, 10 months

[Serusers] Richiesta consenso - Consent request

by mail＠componentsengine.com

Salve, con la presente Vi richiediamo il consenso ad inviarVi materiale informativo, inerente a software e metodologie per la gestione e realizzazione di cataloghi delle parti di ricambio e la loro distribuzione su Internet, Cd-rom e carta. Utilizzi il link sottostante per autorizzare l'invio di materiale informativo. Consento l'invio di materiale informativo <http://www.componentsengine.it/site/page.wplus?ID_COUNT=abilitazionemai llist&LN=1&IDM1=HMArray5_2&IDM2=abilitazione mail list&IDM3=1&@@$$$CHEKUEMAIL@@@@=serusers@lists.iptel.org&PROM_ID=05_12_03&TRAC K_ID=588097&@@$> Una volta dato il consenso potrà esercitare il suo diritto di non ricevere più materiale informativo in qualsiasi momento, tramite la sezione "mail list" <http://www.componentsengine.it/site/page.wplus?ID_COUNT=home&LN=1&IDM1= HMArray1&IDM2=home&IDM3=0&T> presente nel sito internet Per maggiori informazioni consultate il sito internet <http://www.componentsengine.it/site/page.wplus?ID_COUNT=home&LN=1&IDM1= HMArray1&IDM2=home&IDM3=0&T> . Good morning, We ask you for your consent, in order to send you the documentation about the software and the methodologies for the management and the production of the spare parts catalogues as well as its distribution over Internet, CD Rom or paper. Use the link here below, to allow this transmission <http://www.componentsengine.it/site/page.wplus?ID_COUNT=abilitazionemai llist&LN=2&IDM1=HMArray5_2&IDM2=abilitazione mail list&IDM3=1&@@$$$CHEKUEMAIL@@@@=serusers@lists.iptel.org&PROM_ID=05_12_03&TRAC K_ID=588097&@@$> . Once you give your consent, you can enforce your right, in order to receive no documentation in any time by the section "Mail list" <http://www.componentsengine.it/site/page.wplus?ID_COUNT=home&LN=2&IDM1= HMArray1&IDM2=home&IDM3=0&T> in the Internet Website. For more information visit the Internet Website <http://www.componentsengine.it/site/page.wplus?ID_COUNT=home&LN=2&IDM1= HMArray1&IDM2=home&IDM3=0&T> .

20 years, 10 months

[Serusers] nathelper

by Federico R. SANCHEZ Y NAPAL

Hello: I'm using a NATed UA to terminate a call in the PSTN using a Cisco 5300. I want to use the RTP relay capability. In the script, I wrote: ... force_rtp_proxy(); rewritehost("200.60.XXX.XXX"); forward("200.60.XXX.XXX", 5060); ... Where 200.60.XXX.XXX is the GTW IP. I get an error in the log: "/usr/sbin/ser[7039]: ERROR: extract_mediaip: no 'c=' in SDP" The SIP Message sent to de GTW has a wrong content-length header, so the GTW rejects it. Here I copy the message: INVITE sip:2345674284618@200.60.30.24 SIP/2.0 Record-Route: <sip:2345674284618@200.32.43.55;ftag=673501644;lr=on> Via: SIP/2.0/UDP 200.32.43.55;branch=0 Via: SIP/2.0/UDP 200.68.54.46:5060;rport=25127;branch=z9hG4bKFE5D22D6B6C4407884 2396149F64C3DE From: 1001 <sip:1001@200.32.43.55>;tag=673501644 To: <sip:2345674284618@200.32.43.55> Contact: <sip:1001@200.68.54.46:25127> Call-ID: 107D459F-7402-480D-988A-FC87455E1E5D(a)192.168.0.15 CSeq: 31862 INVITE Max-Forwards: 69 Content-Type: application/sdp voip-gtw04# User-Agent: X-Lite build 1101 Content-Length: 315296 v=0 o=1001 23178458 23178458 IN IP4 200.68.54.46 s=X-Lite c=IN IP4 200.32.43.55 t=0 0 m=audio 35052 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active I would appreciate any help about this. Best Regards,

20 years, 10 months

Re: [Serusers] SIP server compatibility

by Jiri Kuthan

At 07:40 PM 1/22/2004, Annie Sasidar wrote: >Hi, > I am working on windowsce.net4.2. The voip architecture in this OS is >based on SIP (RFC2543). I have been trying to register on the Linux SIP >server running ser using a Provisioning.xml file which has the details >required. I am getting the error " Unable to register, check server >settings". I am running the ser-0.8.12 version on the linux box which is >based on RFC 3261. Do you think there is a compatibility issue here? Does >ser allow clients which are based on older RFC's to register? SER registrar handles registrations from both 2543 and 3261 clients -- the error you are seeking must be somewhere else. -jiri

20 years, 10 months

[Serusers] Problem with 0.8.13-dev-23-merged

by giAndrea

Hi when I start this version (0.8.13-dev-23-merged today snashots) I can see in /var/log/messages that ser start correctly but when I try to check server status with serctl moni or simply by connect with MSN ser doesn't work. [cycle #: 1; if constant make sure server lives and fifo is on] I've tried to find some error on log but I didn't found nothing. I'm using a ser.cfg that correctly works with 0.8.12 Any idea? thanks, Andrea ------------------------------------------------------- Andrea Girardi mobile +39 347 9624051 sip address: giandrea(a)sip.quellidelpirana.it http://www.giandrea.com

20 years, 10 months

[Serusers] Looking for a SIP<>H323 Proxy

by Arnd Vehling

Hi, can anyone suggest a Unix based (Linux or freebsd) SIP<>H323 translator? We need to terminate calls with a business partner using H323 so our setup would look like this: SIP Client <> SIP-PROXY <> SIP/H323Proxy <> H323-GK <> PSTN Vovidas siph323csgw cant do this AFAIK so were looking for an alternative. If someone is able to enhance Vovidas siph323csgw so it can talk to gatekeepers we would consider paying for it and publish the code under GPL. best regards, Arnd

20 years, 10 months

[Serusers] why isn't xlog logging?

by Alan Crosswell

I must be doing something really dumb: if (uri==myself) { # allow *all* my aliases to match if (method=="REGISTER") { xlog("L_INFO", "REGISTER %is: %tu\n"); if (!www_authorize("columbia.edu", "subscriber")) { www_challenge("columbia.edu", "0"); break; }; save("location"); # save in usrloc break; }; Nothing shows up in syslog for the xlogs but other syslogging works: Jan 21 14:23:51 ren ser: Aliases: ren.cc.columbia.edu:5060 localhost:5060 localh ost.localdomain:5060 columbia.edu:* Jan 21 14:23:51 ren ser: Listening on Jan 21 14:23:51 ren ser: 127.0.0.1 [127.0.0.1]:5060 Jan 21 14:23:51 ren ser: 128.59.39.127 [128.59.39.127]:5060 Jan 21 14:23:51 ren ser: Aliases: ren.cc.columbia.edu:5060 localhost:5060 localh ost.localdomain:5060 columbia.edu:* Jan 21 14:23:51 ren ser: ser startup succeeded Jan 21 14:24:09 ren /usr/sbin/ser[23883]: ACC: call missed: method=INVITE, i-uri =sip:43754@columbia.edu;user=phone, o-uri=sip:43754@128.59.59.242:5060;user=phon e, call_id=00036baa-c49ce019-38332009-0bf6c446(a)128.59.31.170, from="alan" <sip:a lan(a)columbia.edu>;tag=00036baac49c014725d2d806-0cc644f0, code=487 Request Termin ated version is ser-0.8.12 /a

20 years, 10 months

[Serusers] Call for participants: SIP<>H323 Gateway

by Arnd Vehling

Hello, as the current versions of Vovidas siph323csgw doesnt support connections to H323 Gatekeepers i.e.: SIP UA <> SIP Proxy <> SIPH323csgw <> H323 Gatekeeper <> H323 Gateway we would like to enhance the software with that functionality and make it open source. Right now we have someone who is willing to programm the neccessary features into the siph323csgw for about EU 1000,-. Therefore i am asking who would be interested in donating to the project so we can all have a fully functional, industrial strength SIP to H323 converter. If we get like 4 more "donators" the total cost per participant would be EU 200,- (thats approx USD 215,- atm) Please contact me via email if interested. best regards, Arnd

20 years, 10 months

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sr-users January 2004