sr-users February 2004

sr-users@lists.kamailio.org

144 participants
382 discussions

[Serusers] Cisco As5300 SIP/ISDN Interworking
by Wolfgang Platzer 19 Feb '04

19 Feb '04

Hi, does anybode know a SIP/SDP field that can be used to set the Bearer Capability to Speech or 3.1 kHz on the outgoing Setup message? We need this for FAX calls because some devices do not accept speech calls. regards, wolfgang

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[Serusers] monitoring SER
by Nancy Rogers 19 Feb '04

19 Feb '04

Hi Alex, Was wondering if you ever got Nagios to monitor a SER proxy? (Serusers from Nov). Thanks. Wes.

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[Serusers] LDAP authentication
by GUSTAVO GARCIA BERNARDO 19 Feb '04

19 Feb '04

Hi, I would like to test the LDAP authentication support in SER. What module should i use? Are there any sample configuration? Thx, best regards. G.

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[Serusers] multiple chained media proxy
by Adrian Georgescu 18 Feb '04

18 Feb '04

>- if there are two or more Media Proxies in a row in your session, you > might end up in a deadlock situation, resulting in no media transmitted > (general issue). Bernie, You can circumvent this in the stable version by adding/checking an extra header: if (method=="INVITE") { xlog ("L_INFO","-------------------------"); if (search("^P-Behind-NAT.*Yes.*") || search("^P-RTP-Proxy.*YES.*")) { xlog ("L_INFO","Already one Media Proxy in the path"); } else { xlog ("L_INFO","Use SerMediaProxy - Incomming call leg"); append_hf("P-Behind-NAT: Yes\r\n"); force_rtp_proxy(); } t_on_reply("1"); }; In unstable Maxim did something already in this respect and I will provide an update soon to have SerMediaProxy work out of the box. -- Adrian Hi! I have such a setup because in my scenario only the Media Proxy has the rights to receive connections coming from the Internet. Direct traffic to all other hosts (UAs) is blocked. It might be that you have to insert one more line, i.e. setflag(1) (don't ask me why...:-) ) Below a cut-past from my testing configuration, which seems to work fine ---- [...] # force RTP-proxy if (method=="INVITE") { record_route(); setflag(1); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } # RTP-proxy # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { if (status=~"[12][0-9][0-9]") force_rtp_proxy(); } --- Be prepared on the following possible issues: - Clients, which do not support symetric media (sender port==receiver port) will probably not work. (Most of those anyway do not work behind NAT.) - If you have more than one media (e.g. Audio and Video) in the same sesssion, only one media (the first m= line) is proxied correctly. The other media probably won't work (general issue). - if there are two or more Media Proxies in a row in your session, you might end up in a deadlock situation, resulting in no media transmitted (general issue). Have fun! T: Bernie On Wed, 18 Feb 2004, CURRO_DOMINGUEZ wrote: > Hello, > > thanks all the list for help me (and others) with nathelper and rtpproxy. > > I want to test rtpproxy on my network, but without nat translation. It > means, udp streams have to go through my server. > > I'm not sure what functions from nathelper module I have to use. My idea > is something like this: > > if (method=="INVITE"){ > force_rtp_proxy(); > t_on_reply("1"); > } > .... > > onreply_route[1]{ > force_rtp_proxy(); > } > > I don't know whether this code is enough, or I have to use another > functions like fix_nated_contact() or fix_nated_sdp() to rewrite Contact > an SDP headers. > > Thanks for your help. > > > > > _______________________________________________ > Serusers mailing list > Serusers at iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Finding the right hardware to work with SIP
by Helder Rogerio [MICROREDE] 18 Feb '04

18 Feb '04

Could someone tell me if Cisco ATA 186 should work ok with SER? Or someone tried it even if direct device-to-device voice vpn? The data about this gateway i've quoted it below. Any suggestion about equipments like routers, gateway to regular phones (analogue), PBX gateways to VoIP would be appreciated because we have bought two Draytek Vigor VoIP enabled routers with VPN support and now the fabricant tolds that the problem that we have is because of some firmware problems to be resolved in March by the time of Cebit event. :-( Thanks in advance Helder Voice-over-IP (VoIP) protocols H.323 v2 H.323 v4 SIP (RFC 2543 bis) MGCP 1.0 (RFC 2705) MGCP 1.0/network-based call signaling (NCS) 1.0 Profile MGCP 0.1 SCCP Voice codecs1 G.729, G.729A, G.729AB2 G.723.1 G.711a-law G.711µ-law Cumprimentos / Best regards, Helder Rogério __________________________________________ Microrede - Tecnologias de Informação, Lda. Rua Lopes, 55 - C/V E 1900-297 Lisbon Portugal Tel. 21 8148372 Fax. 21 8127158 Web: http://www.microrede.com *** « There are only two types of people in the world, those who have lost data and those who will. » -- Richard Nixon

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[Serusers] mysql trouble
by Mark Chehovsky 18 Feb '04

18 Feb '04

Hi I had installed SER 0.8.11 on my FreeBSD 4.8 and it worked well. After changing IP address on machine SER dont start. In syslog i have : Feb 18 12:14:01 voip ser: connect_db(): Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) Feb 18 12:14:01 voip ser: db_init(): Error while trying to connect database Feb 18 12:14:01 voip ser: mod_init(): Error while connecting database Feb 18 12:14:01 voip ser: init_mod(): Error while initializing module usrloc database ser exists and user ser with password heslo has unlimited access to all tables. Any ideas ? :) Mark

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[Serusers] Voicemail (Once more..)
by Atle Samuelsen 18 Feb '04

18 Feb '04

Hey guys.. anybody know why im getthing this error? Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command vm_reply is not available Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command must have at least 3 chars Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command must begin with :: 0000728447A2667A Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command must begin with :: Content-Type: application/sdp Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command must begin with :: v=0 Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command must begin with :: s=session Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command must begin with :: t=0 0 Feb 18 11:43:32 voip ser[16117]: ERROR: fifo_server: command must begin with :: a=rtpmap:0 PCMU/8000 It happens when I trigger the voicemail.. - Atle

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RE: [Serusers] eliminating 483 errors
by Ticknor.Scott＠ic.gc.ca 18 Feb '04

18 Feb '04

> -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, February 18, 2004 10:09 am > To: Ticknor, Scott: DGSE > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] eliminating 483 errors > > > I think the alias must be in one line, or only the last line > will be used! > > from your dump: the proxy forwards the invite to the second phone - > which does not answer. There is no 483 in your dump. > > klaus i don't understand it. serctl moni shows 483 whenever i use the Cisco [sip fw v6.1] phones. i changed the display name on the phones to match the phone number, just in case that was problematic. the call as shown in the dump [that is only part of the dump, the full file is 48 pages in word :/] results in: a) phone A calls phone B normally b) phone A is hung up, phone B sees the hangup and disengages [i can see the screen on phone B change as if call was cancelled] c) phone B immediately starts ringing again, thinking it is receiving the same call again. d) serctl moni notes 483 errors, and the 'Failure' count goes up as well. X-Lite, MSN, and Kphone clients seem to work ok. a tail of /var/log/messages shows the same message repeating [a summary from memory]: sl_reply won't send reply for Ack check out the ser.cfg, this is the one in use right now. it is nearly stock, just a couple edits that shouldn't make much difference!? st

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RE: [Serusers] eliminating 483 errors
by Ticknor.Scott＠ic.gc.ca 18 Feb '04

18 Feb '04

i attached one ser.cfg file. and a dump file from ngrep. it looks like a bunch of INVITES. the Cisco phone B rings, but if i hang up Cisco phone A, B continues to ring and ring... and 483 errors every few seconds, what a mess. > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, February 18, 2004 9:22 am > To: Ticknor, Scott: DGSE > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] eliminating 483 errors > > > post your ser.cfg and a dump of the packets that causes the loop. > > Klaus > > Ticknor.Scott(a)ic.gc.ca wrote: > > hello List, > > i am sadly aware that the 483 problem is covered in the > Admin guide & faq. > > the solution as i read it is to replace if (uri==myself) > with some regexp > > for the sip proxy. but this is apparently not a proper > implementation, from > > what i have read on the list. furthermore, the if > (uri==myself) expression > > is further down in the logic, below the check for maxfwd > that results in the > > 483. if have set SIP_DOMAIN and added aliases to ser.cfg. i > am sure i am > > missing something small and simple... > > my clients are cisco 7960 with sip fw v6.1. > > > > thanks for reading, > > > > scott t > > DSi > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

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[Serusers] rtpproxy but no nat translation needed
by CURRO_DOMINGUEZ 18 Feb '04

18 Feb '04

Hello, thanks all the list for help me (and others) with nathelper and rtpproxy. I want to test rtpproxy on my network, but without nat translation. It means, udp streams have to go through my server. I'm not sure what functions from nathelper module I have to use. My idea is something like this: if (method=="INVITE"){ force_rtp_proxy(); t_on_reply("1"); } .... onreply_route[1]{ force_rtp_proxy(); } I don't know whether this code is enough, or I have to use another functions like fix_nated_contact() or fix_nated_sdp() to rewrite Contact an SDP headers. Thanks for your help.

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sr-users February 2004