sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

RE: [Serusers] SER with asterisk as voicemail problem... please help!

by Quang Minh PHAN

Hi, It seems to be a problem in Asterisk configuration, not in SER's. Showing your asterisk's extensions.conf file may help. Regards, Minh -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Ryan Pagquil Envoyé : mardi 20 septembre 2005 09:50 À : serusers(a)lists.iptel.org Objet : [Serusers] SER with asterisk as voicemail problem... please help! Hi, I configured SER to forward certain calls to asterisk for the voicemail functionality. But when I forward the SIP request to Asterisk my xlite softphone prompts "403", "Forbidden". This is my ser.cfg part where i forward the certain uri to the asterisk. if (uri=~"^sip:1[0-9]{3}@sip.philonline.com.*$") { strip(1); rewritehostport("202.84.24.47:5060"); t_relay(); break; }; Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 2 months

[Serusers] need help

by julien bossart

Hi all, I am new to this forum. Today, i download ser package, then i install in my fedora core 4. Everything is ok. Next, i start ser and use serctl add to add account to ser server. Then i use sjphone to connect to ser server. Sjphone try to send REGISTER to ser server. I use ethereal to get the package. I see that ser server responds ICMP host unreacheable all the time. So, REGISTER is failed. Can you give me some suggestion? Thank you for that.

19 years, 2 months

[Users] Compilation error on Mac OS

by Amos Nungu

I'm installing SEMS-0.9 on Mac OS (Darwin 8.2.0). upon running the make all command, I got an error as shown below ------------------------------------------------- # make all make[1]: Nothing to be done for `deps'. g++ -o sems AmApi.o AmAudio.o AmCmd.o AmConfig.o AmEventQueue.o AmIcmpWatcher.o AmMail.o AmPlugIn.o AmRequest.o AmRtpPacket.o AmRtpPacketTracer.o AmRtpScheduler.o AmRtpStream.o AmSdp.o AmServer.o AmSession.o AmSmtpClient.o AmThread.o AmUtils.o EmailTemplate.o SemsConfiguration.o SerClient.o SerDBQuery.o log.o sems.o -lm -ldl -lsocket -lnsl -lpthread /usr/bin/ld: can't locate file for: -lsocket collect2: ld returned 1 exit status make[1]: *** [sems] Error 1 make: [all] Error 2 (ignored) ------------------------------------------------------- Any idea on how to go about it will be great. Regards

19 years, 2 months

[Users] SER + MediaProxy or rtpproxy

by Frank Kostin

Hi All, Using Onsip SER with Mediaproxy --no-fork I can see caller/called/relayed. This is OK for local calls only. With or without mediaproxy I have audio both directions. When I call to /from another network the phone rigning but no audio. Can I relay /forward by IP address using mediaproxy and eventually redirect media streams throught a 3rd party ? Thanks, Frank --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

19 years, 2 months

[Serusers] Problem redirecting to voicemail

by Chris St Denis

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail. GATEWAY---PROXY---FEATURE | | UA For simplicity, hostnames and IPs replaced with the above names. USERNAME, DSTNUM and SRCNUM also used to replace the UA's username, the source number of the call, and the destination number of the call. The basic SIP dialog goes: Gateway invites proxy Gateway invites UA UA replies 180 Ringing. Transaction times out and drops to failure route PROXY invites FEATURE server (INVITE sip:*voicemail-busy-2002006@ FEATURE:5060 SIP/2.0.) PROXY cancels UA FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY UA tells PROXY '487 Request Terminated.' FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY ... For some reason, the asterisk gateway doesn't seem to be ACKing the 200 ok. I don't see any new invite or reinvite going to the gateway which I think may be what is confusing it. Ser.cfg excerpt ## # User did not answer phone, or could not connect, or is on the phone and does not use call waiting. ## failure_route[3] { if(isflagset(10)) { if(t_check_status("486")) { if (!subst_user('/^/*voicemail-busy-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail Busy\n"); } else { if (!subst_user('/^/*voicemail-noanswer-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail No answer\n"); } rewritehostport("FEATURE:5060"); append_branch(); t_relay(); } } Ngrep from PROXY's point of view http://pastebin.ca/23469 Ngrep from GATEWAY's point of view http://pastebin.ca/23470 Ngrep from FEATURE's point of view http://pastebin.ca/23471

19 years, 2 months

[Serusers] Can SER determine NAT type?

by Chris St Denis

I want to use STUN as a primary nat solution, with the nathelper or mediaproxy module doing rewrites as a secondary/backup solution. However, this will not work for symmetric NATs. I want to proxt the rtp data from these (and only these) kind of nat. Can SER determine that? My client adds a Warning: 399 spa "Restricted Cone NAT Dectected". Header to the register message (determined by stun), but that looks proprietary and not something I can expect to be the same from other clients so I'm not sure I want to header match on that.

19 years, 2 months

[Serusers] isntalling SER on Red hat linux

by rajesh volluru

hi Can we install SER on Red Hat linux. If yes, then please send me the instructions. Also, I get Transaction unavailable exception when I try to communicate through Sip Communicator. Does installing SER solve that problem? Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

19 years, 2 months

Re: [Serusers] COnfiguring a PSTN number with SER

by Iqbal

in ur messages file, i have u traced to see where in ur config its getting stuck, i mean when it gos through with the INVITE when it does alias tabel lookup is it matching anything, also look in mysql log file to see what info it is selecting upon. what is the format of your entry in aliases, it will map your username to alias, wheras you want number mapped to username i think iqbal Ashutosh Kumar wrote: >Hi, > I assigned the DID to the user but serctl add alias, and the entry >got updated in ser.aliases table. > But when I call the number from pstn, the registered user does not get >the call. The nextone contacts the SER, but SER responsds to it with a "478 >unresolvable desintation". > Following is the ngrep excerpt. > >U PSTN_GW_IP:5060 -> SER_IP:5060 >INVITE sip:@SER_IP SIP/2.0. >Max-Forwards: 70. >Session-Expires: 3600;Refresher=uac. >Supported: timer. >. >v=0. >o=msw1 1234 1929 IN IP4 x.y.z.101. >s=sip call. >c=IN IP4 x.y.z.101. >t=0 0. >m=audio 27468 RTP/AVP 0 8 3 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:8 PCMA/8000. >a=rtpmap:3 GSM/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-16. >a=silenceSupp:off - - - -. > >## >U SER_IP:5060 -> PSTN_GW_IP:5060 >SIP/2.0 100 trying -- your call is important to us. > > >############# >U SER_IP:5060 -> PSTN_GW_IP:5060 >SIP/2.0 478 Unresolvable destination (478/TM). > >Relevant code from ser.cfg is as follows: > >... >... >modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") >modparam("usrloc", "db_mode", 2) > > > if (uri==myself) { > > if (method=="ACK") { > route(7); > break; > } else if (method=="CANCEL") { > route(3); > break; > } else if (method=="INVITE") { > route(3); > break; > } else if (method=="MESSAGE"){ > route(4); > break; > } else if (method=="REGISTER") { > route(2); > break; > }; > > lookup("aliases"); > if (uri!=myself) { > route(5); > route(1); > break; > }; > > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > break; > }; > > >I don't know where I am wrong. > >Regards, >Ashutosh > > > > > >-----Original Message----- >From: Iqbal [mailto:iqbal@gigo.co.uk] >Sent: Tuesday, September 20, 2005 10:35 PM >To: Ashutosh Kumar >Cc: 'Andreas Granig'; serusers(a)lists.iptel.org >Subject: Re: [Serusers] COnfiguring a PSTN number with SER > >just alias that number to a voip username which is bound to that user, >so when 123456789 is dialed nextone sends it to ser, ser looks up in >aliases, sees the match and sends it to that user > >Iqbal > >Ashutosh Kumar wrote: > > > >>Hi Andreas, >> Thanks for the reply, I was thinking of another approach to do this, >>tell me if I am wrong. >> May I use the 'phone' field in subscriber table to run an external >>sql script to map the DID (phone) with a username, and then rewrite the >> >> >uri, > > >>and then forward the call to that user? >> >>Regards, >>Ashutosh >> >> >>-----Original Message----- >>From: Andreas Granig [mailto:andreas.granig@inode.info] >>Sent: Tuesday, September 20, 2005 6:22 PM >>To: Ashutosh Kumar >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] COnfiguring a PSTN number with SER >> >>Ashutosh Kumar wrote: >> >> >> >> >>> I have a DID number to which I want to assign to a user in >>>SER, the call will come through a NExtone. How / what changes I have to >>>make in ser.cfg to arrange that whenever a call comes on that number on >>>nextone, that call should be forwarded to the sip-user bound to that >>> >>> >>> >>> >>number. >> >> >> >> >>>How can I use aliases in this scenario. >>> >>> >>> >>> >>Check the PSTN chapter of the Getting Started document at http://onsip.org >> >>Andy >> >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >>. >> >> >> >> >> > > > >. > > >

19 years, 2 months

[Serusers] COnfiguring a PSTN number with SER

by Ashutosh Kumar

Hi, I have a DID number to which I want to assign to a user in SER, the call will come through a NExtone. How / what changes I have to make in ser.cfg to arrange that whenever a call comes on that number on nextone, that call should be forwarded to the sip-user bound to that number. How can I use aliases in this scenario. Please guide. Regards, Ashutosh Kumar

19 years, 2 months

[Serusers] obsfuctaing ser.cfg

by Iqbal

is this possible other than just removing the spaces and all. Iqbal

19 years, 2 months

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sr-users September 2005