sr-users June 2005

sr-users@lists.kamailio.org

241 participants
595 discussions

[Serusers] ser don't start
by harry gaillac 04 Jun '05

04 Jun '05

Hello ser users, I've sent many mails about starting ser. my problem is solved . ser sent to syslog: auth_db:str_fixup: Invalid table version (use ser_mysql.sh reinstall) auth_db module did not find table "subsciber" below. if (!www_authorize("nxs.yi.org", "subsciber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; I fix table to "subscriber" ! Thanks ser users for help Regards Harry _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

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RE: [Serusers] Mediaproxy in LAN (?)
by Giudice, Salvatore 04 Jun '05

04 Jun '05

I am trying to route a call from a non-routable 10.x.x.x network to a phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN to SIP from my asterisk to an x-lite I found some references indicating that I should call: if ( src_ip = IP_GATEWAY) { force_rport(); fix_contact(); use_media_proxy(); }; Supposedly, you need the force_rport and the fix_contact to put media proxy in the middle of the two rtp sessions, but I have et to see this actually work. Has anyone been able to pass a call through media proxy from a GW on a non-routable to a routable network, or even between two ip's on the same network as the mediaproxy? If so, please share your configs. -----Original Message----- From: bogdan aux [mailto:aux1d@yahoo.com] Sent: Friday, May 13, 2005 6:16 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Mediaproxy in LAN (?) I want to route RTP packets between 2 computers in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly from one computer to another. My guess is mediaproxy makes a test on the IPs (they are 192.168.147.2 and 192.168.147.3) and refuses to proxy the call. Is mediaproxy working just between different networks or it's an error in my configuration file? Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] CDRTool
by Mr Greg Plater 03 Jun '05

03 Jun '05

Is anyone using the CDRTool from AG I have installed and configured it for testing. Its missing the Rating engine etc. Anyone know the cost of a Lic with the rating engine ?

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[Serusers] Asterisk and SER
by Ronald Wiplinger 03 Jun '05

03 Jun '05

I read about SER and I have installed Asterisk on one machine. Unfortunately this machine is broken and the replacement will take longer as I wished. I have a P-4, 1GHz with 1 GB RAM, which was planned to be used as SER anyway. I would like to use this downtime of the Asterisk box to install SER. I understand that SER and Asterisk can work together, while SER makes connection, Asterisk will add features. How do I do that efficiently? 1. SER needs a public IP, does then Asterisk need a pulic IP, or can it just be on the internal LAN? 2. Now the password for each phone is made in the SIP and IAX config files. How can I do that than in SER? If I use reatime, can I use the same database? 3. A user can make a phone call to the users on our system (Asterisk domain) and via gateways. We use ASTCC for billing. How can I use this together with SER. I am not sure if I understand it correct, but would for internal calls (calls without features) be handeled directly from SER, while if the user need a feature, than he would be transfered to Asterisk? How can I setup multiple Asterisk boxes with one SER, but all Asterisk machines share the same dial plan (realtime) I wanted to read more first, but the down time of my * forces me to do a SER setup faster, so that my users can quickly (by office hours Monday) at least make internal calls. Any hint is welcome ;-) bye Ronald

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[Serusers] SER 0.9.2 & Route & Analog FXO Problems :(
by Ozan Blotter 03 Jun '05

03 Jun '05

Dear Friends, I know this is annoying to post nearly same things but still no answer :( Is it really so hard or can anybody tell me why this is not working. There are : ATA 1 : 12345 (registered and can call 54321, no problem) ATA 2 : 54321 (registered and can call 12345, no problem) ANALOG FXO : 201 (registered as 201, for test purposes i only registered one port of it) With Brekeke's Ondo SIP Proxy, i created a simple dial-plan and it worked, even i can add more prefixes no problem. But in SER i cannot :( I'm using ver 0.9.2. Normally it's like when i want to call a GSM, proxy simply adds a prefix 201 which is same with registered sip number for regarding analog port and forwards it (looks like sip:20105353490056@192.168.1.35) to Gateway and Gateway simply strips 201 from it and dials 05353490056, yes it connects. But what this same does not work on SER, any ideas because it's very boring to play all day and night ? Thanks.... >>>>>>>>>> minimal SER.CFG >>>>>>>>>> debug=3 fork=yes log_stderror=no check_via=no dns=no rev_dns=no listen=192.168.1.10 port=5060 children=4 fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if (uri=~"^sip:053[0-9]*@.*") { prefix("201"); rewritehostport ("192.168.1.35:5060"); # forward ("192.168.1.35:5060"); t_relay_to_udp("192.168.1.35","5060"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; } >>>>>>>>>> call from 12345 to 54321 (SIP 2 SIP Call) >>>>>>>>>> U 192.168.1.10:5060 -> 192.168.1.201:5060 INVITE sip:54321@192.168.1.201:5060 SIP/2.0..Record-Route: <sip:192.168.1.1 0;ftag=xITO2cDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9hG4bK233f .8f8b9d41.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKba4b3e1d3eed 9c7c..Max-Forwards: 16..To: <sip:54321@192.168.1.10>..From: <sip:12345@192. 168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID: 911D1B912D91212(a)192.168.1.20 0..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..Session-Expires : 100;refresher=uac..Supported: timer..Content-Type: application/sdp..Conte nt-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-..c=IN IP4 192. 168.1.200..t=0 0..m=audio 2142 RTP/AVP 18 18..a=rtpmap:18 G729/8000..a=send recv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.200:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKba4b 3e1d3eed9c7c..To: <sip:54321@192.168.1.10>;tag=xETOxUDMxED..From: <sip:1234 5(a)192.168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID: 911D1B912D91212(a)192.16 8.1.200..CSeq: 1 INVITE..Record-Route: <sip:192.168.1.10;ftag=xITO2cDMxID;l r=on>..Content-Length: 0.... >>>>>>>>>> call from 12345 to 05353490056 (GSM Call) >>>>>>>>>> U 192.168.1.10:5060 -> 192.168.1.200:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.16 8.1.200:5060;branch=z9hG4bK31bc676961f91e18..To: <sip:05353490056@192.168.1 .10>..From: <sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 8 76D1B912D91212@192.168.1.200..CSeq: 1 INVITE..Server: Sip EXpress router (0 .9.2 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.10:5060 "Nois y feedback tells: pid=4587 req_src_ip=192.168.1.200 req_src_port=5060 in_u ri=sip:05353490056@192.168.1.10 out_uri=sip:20105353490056@192.168.1.35:506 0 via_cnt==1".... # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.200:5060 SIP/2.0 200 ok -- no more pending branches..Via: SIP/2.0/UDP 192.168.1.200: 5060;branch=z9hG4bK31bc676961f91e18..To: <sip:05353490056@192.168.1.10>;tag =2f9bfc2acf470ceacf4efdebbaa289b4-026c..From: <sip:12345@192.168.1.10;user= phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D91212@192.168.1.200..CSeq: 1 CAN CEL..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length: 0..Wa rning: 392 192.168.1.10:5060 "Noisy feedback tells: pid=4586 req_src_ip=19 2.168.1.200 req_src_port=5060 in_uri=sip:05353490056@192.168.1.10 out_uri=s ip:20105353490056@192.168.1.35:5060 via_cnt==1"....

2 1

[Serusers] auth_db module table parameter?
by Cameron Beattie 03 Jun '05

03 Jun '05

I want SER to use another application's database but can't see a way to tell the auth_db module which table to use. I can see the paramaters for the host/database name, and the column names. But how do I tell SER which table to use or does it only ever use subscriber? Regards Cameron

2 2

[Serusers] CDR not being recorded at ACC table
by Felipe Martins 03 Jun '05

03 Jun '05

Hi everybody, I want to test my billing system based on the CDR data I've recorded at ACC table, but there is a problem, the data is not being recorded as I thought. I don't know why, when I go directly through mysql and make a "select * from acc;" it gives me: Empty set (0.00 sec) I don't know why the CDR date is not being recorded, I looked for it at the forum and google, and found something about mysql installation. It seems that I haven't compiled mysql with ACC support. Is that right ? Did someone have the same problem already ? Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

2 1

[Serusers] avpops / failure route
by Martin Koenig 03 Jun '05

03 Jun '05

Hello, is it possible to use avp_write on the responses in failure route? If yes, Contact header field could be evaluated and pushed into RURI, then routing via route(x) to a new destination and finally solve the 3xx forwarding issue. Regards, Martin

2 3

[Serusers] ngrep of call transfer
by Iqbal 03 Jun '05

03 Jun '05

Hi Does anyone have a dump of what a succesful call transfer for the following two scenarios looks like PSTN <---> PHONE A PHONE A -----> PHONE B PSTN <----> PHONE B basically A transfer the call to B the next scenario is the same but instead of PSTN , its between IP only PHONE A <---> PHONE B PHONE B ---> PHONE C PHONE A <---> PHONE C tks Iqbal

1 0

[Serusers]
by Grigory Fishilevich 03 Jun '05

03 Jun '05

hi guys, is it possible to use a global phone book with ser? i found no features like this in sip rfcs, but this is imho very impotant thing for companys etc. do anybody know an way to implement a global phone book for all users of one ser? bg, grigory

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sr-users June 2005