sr-users May 2004

sr-users@lists.kamailio.org

154 participants
333 discussions

[Serusers] Video-calls
by Roberto Hiribarne Guedes 06 May '04

06 May '04

Hi, I need to know of any SIP cmpatible free video client for windows ... Or if I have to use NetMeeting and put an H323/SIP gateway in the middle of client and proxy ... What I need is something like X-Lite soft phone ... with video suport ... For Windows ... and free ... Thanks, R.

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[Serusers] t_on_failure and accounting
by Zeus Ng 06 May '04

06 May '04

Hi all, Can anyone suggest a better logic for me. What I want is to account "Busy" or "DND" as missed calls and forward the request to voicemail. In my config (attached), I have a t_on_failure block which will relay the request to voicemail system. The trouble is, once the request is relayed, SER will not treat this as a missed call. Instead, SER will treat it as a successful call with the duration of the voicemail. It will also account two BYEs. I have tried to send a 'redirect to voicemail' reply to UA in the failure block. But than, the account reason becomes a "300" response, not "4xx" response. failure_route[2] { xlog(L_NOTICE", "Callee busy or DND, forwarding to voicemail\n"); revert_uri(); lookup("aliases"); subst_uri('/(sip:)(.*)@(.*)/<\1voicemail-\2@\3>/i'); t_reply("300", "Redirect to voicemail"); } Unless I can use acc_db_request() to do manual accounting or sl_send_reply("300", ...) to send the redirect in the failure block, I don't know to change the logic to get what I want. Any idea? Zeus Ng

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[Serusers] mod_init: bind_db failed..
by Andy Pyles 06 May '04

06 May '04

Hi I'm running trying out latest CVS checked out today with the following configuration and am having the following error: May 6 00:56:54 hostname /usr/local/sbin/ser[30030]: ERROR:acc:mod_init: bind_db failed...did you load a database module? May 6 00:56:54 hostname /usr/local/sbin/ser[30030]: init_mod(): Error while initializing module acc However I can connect to mysql just fine, from the command line. In addition I just ran ser_mysql.sh create. Anyone know I am getting this error? Attached is my ser.cfg with no modifications ( this is stock from ser.cfg.m4) regards, Andy

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[Serusers] Ipv6 to PSTN setup
by Lambros Lambrinos 06 May '04

06 May '04

Hi, I've been experimenting with the following setup: ipv6 ua --- ser (v6) --- msp gateway --- ser (v4) --- cisco 3640 pstn gateway --- cisco ip-phone Everything works fine with the IP phone but routing the call to the PSTN gateway causes it to return a 400 Malformed/Missing Request in response to the packet forwarded to it from ser(v4) If i use the ip-phone to make a call via the pstn gateway (i.e. involving only the V4 SER) everything is fine. This makes me believe that the most likely cause of the crash is that the PSTN gateway can not understand the IPv6 address in the Via: or the Record-Route: header fields If that is the case, where do people think would be the best place to remove the 'offending' fields and add them back when the reply comes back from the PSTN gateway ? The SERv4 server or the MSP gateway ? The last packet sent to the PSTN gateway is: Session Initiation Protocol Request-Line: INVITE sip:3007@128.16.67.2:5060 SIP/2.0 Message Header Record-Route: <sip:3007@128.16.234.7;ftag=49F56B1E;lr=on> Max-Forwards: 9 Record-Route: <sip:[2001:630:13:101:2E0:18FF:FE3E:ECB2];ftag=49F56B1E;l r =on> Via: SIP/2.0/UDP 128.16.234.7;branch=z9hG4bK86ba.d7fafbd5.0 Via: SIP/2.0/UDP 128.16.67.73:5062 Via: SIP/2.0/UDP [2001:630:13:101:2e0:18ff:fe3e:ecb2];branch=0 Via: SIP/2.0/UDP [2001:630:13:101:204:23ff:fe0b:dc58] CSeq: 6566 INVITE To: <sip:3007@128.16.234.7> Content-Type: application/sdp From: "Manish Lad" <sip:1006@frostie.cs.ucl.ac.uk>;tag=49F56B1E Call-ID: 1658003513 Subject: sip:1006@frostie.cs.ucl.ac.uk Content-Length: 185 User-Agent: KPhone/3.11 Contact: "NAT Proxy" <sip:proxy@128.16.67.73:5062;realuri=0-ralOcJk1sRAM 6AlMTdrMoAvMTMvMTAmO6IlNDdnMrZbObZaMGIvZGMqODip-bFj-sBk-cQz9WRl> Session Description Protocol Session Description, version (v): 0 Owner/Creator, Session Id (o): username 0 0 IN IP4 128.16.67.73 Session Name (s): The Funky Flow Connection Information (c): IN IP4 128.16.67.73 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 32974 RTP/AVP 0 97 3 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 The reply from the gateway: Session Initiation Protocol Status-Line: SIP/2.0 400 Bad Request - 'Malformed/Missing Record Route' Message Header Via: SIP/2.0/UDP 128.16.234.7;branch=z9hG4bK86ba.d7fafbd5.0,SIP/2.0/UDP 128.16.67.73:5062,SIP/2.0/UDP [2001:630:13:101:2e0:18ff:fe3e:ecb2];branch=0,SIP / 2.0/UDP [2001:630:13:101:204:23ff:fe0b:dc58] From: "Manish Lad" <sip:1006@frostie.cs.ucl.ac.uk>;tag=49F56B1E To: <sip:3007@128.16.234.7>;tag=88098FAC-15AD Call-ID: 1658003513 CSeq: 6566 INVITE Content-Length: 0 Regards, Lambros

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[Serusers] Call drops with 60 sec and Buffer size exceeded message
by Jadylson Bomfim 06 May '04

06 May '04

Hi guys, I was create a script to route some kinds of routes calls to PSTN ( like that ). So after 59 sec. it is droped. Someone can help me ? Thank you all for attention and kindness in advance. Jadylson Bomfim if (uri=~"sip:0[0-9]*@.*") { route(6); break; }; route[6] { setflag(1); if (uri=~"^sip:0[1-9]*@") { strip(1); prefix("01155"); rewritehostport("ip address:5060"); acc_rad_request("Saindo Para PSTN no Brasil"); if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("ip address","5060"); } else { if (uri=~"^sip:00[0-9][0-9]*@") { # ... forward to gateways then; #Cheque se inicia com 00 if (uri=~"^sip:00") { # strip the leading "1" strip(2); prefix("011"); # if you have passed through all the checks, let your call go to GW! rewritehostport("ip address:5060"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("ip address","5060"); }; } else { if (uri=~"^sip:001*@") { # ... forward to gateways then; #Cheque se inicia com 00 if (uri=~"^sip:00") { # strip the leading "1" strip(2); # if you have passed through all the checks, let your call go to GW! rewritehostport("ip address:5060"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("ip address","5060"); }; }; }; }; } Jadylson da Rocha Passos Bomfim Redevox Telecom Uberlandia +55 34 3234-7813 São Paulo +55 11 5055-6888 Móvel +55 34 9103-6854 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.643 / Virus Database: 411 - Release Date: 25/3/2004

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[Serusers] Alias Expired
by Ezequiel Colombo 06 May '04

06 May '04

Hi all, i am new in this list. In my ser-0.8.12-0 (rpm installed) system i have problems with aliases, when create an alias binding between any numerical alias and my alphanumeric user with serctl i note that the Expires field is filled by an negative number like -3221225512. Whe i do a serctl alias show [numeric alias] i receive a "404 No registered contacts found" and the line "Binding '4517','sip:ecolombo@arcotel.net' has expired" in my messages file. The alias is never dumped to the mysql database and if i call to the alias a 404 Not Found is received. ------------------------------------------------- Paste of system commands and results: ------------------------------------------------- [root@myser log]# serctl alias show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0x422b42b8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0x422b2258 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ===/Domain list=== [root@myser log]# serctl alias add 4517 sip:ecolombo@arcotel.net sip:ecolombo@arcotel.net 200 Added to table ('4517','sip:ecolombo@arcotel.net') to 'aliases' [root@myser log]# serctl alias show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0x422b42b8 d_ll { n : 1 first: 0x422b62c0 last : 0x422b62c0 } ...Record(0x422b62c0)... domain: 'aliases' aor : '4517' ~~~Contact(0x422b6300)~~~ domain : 'aliases' aor : '4517' Contact: 'sip:ecolombo@arcotel.net' Expires: -3221225478 q : 1.00 Call-ID: 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 replic : 0 State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0x422b2258 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ===/Domain list=== [root@myser log]# serctl alias show 4517 404 No registered contacts found [root@myser log]# serctl alias show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0x422b42b8 d_ll { n : 1 first: 0x422b62c0 last : 0x422b62c0 } ...Record(0x422b62c0)... domain: 'aliases' aor : '4517' ~~~Contact(0x422b6300)~~~ domain : 'aliases' aor : '4517' Contact: 'sip:ecolombo@arcotel.net' Expires: -3221225512 q : 1.00 Call-ID: 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 replic : 0 State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0x422b2258 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ===/Domain list=== [root@myser log]# tail -2 /var/log/messages May 6 11:52:56 billing ser: ser startup succeeded May 6 11:53:56 billing /usr/sbin/ser[30873]: Binding '4517','sip:ecolombo@arcotel.net' has expired Thanks Ezequiel Colombo ARCOTEL S.A.

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[Serusers] SER pa Module not working ????!
by Oliver Zilken 06 May '04

06 May '04

Hi there, I have got a problem with ser pa module (presence module). When I include it into my ser.cfg ( loadmodule "/usr/local/lib/ser/modules/pa.so" ), I cannot start ser correctly. Here is my syslog: May 6 13:23:05 sip ser: Listening on <<<<<<<<<<<<<<<SNIP>>>>>>>>>>>>>>>>> May 6 13:23:06 sip ser: ser startup succeeded This should mean that ser is running. But when i call "service ser status", it says: ser dead but subsys locked Has anyone a reason for this ? I am planning to include presence display in serweb... Oliver

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[Serusers] serweb
by gaillac harry 06 May '04

06 May '04

hello Logout and Faq don't display in user_management page contrary admin page so users can't logout !! I looked in page.php without succes to solve this problem I use serweb 2004 Harry

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[Serusers] Fw: DB replication
by Andrea Bondavalli 06 May '04

06 May '04

Hi all, in order to achieve reliability I am currently trying to setup two SIP proxy responsible for a single DNS SRV domain. My User Agents do support a DNS fail-over mechanism and they are able to communicate with the secondary Proxy Server when the dialog with the Primary Proxy fails. At this time I am setting up two proxy servers (primary and secondary) both responsible for the same DNS SRV domain. Each Proxy do connect to its local mysql database (located on the same host of the SER) and they are both configured to exchange REGISTERs using the t_replicate() function. Now I have to setup the mysql DB replication and I have some questions: 1) Is the described way of setting up a primary and a secondary Proxy server right ? 2) Which kind of DB replication should I setup ? 3) Do I have to exclude the USRLOC table from the replication or not ? (the USRLOC replication should be guaranteed by the REGISTERs exchange) Thank you in advance, Andrea Bondavalli I.NET S.p.A. - Research & Development mailto:a.bondavalli@inet.it Tel. +39.02.328637232 fax. +39.02.328637702

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[Serusers] mysql accounting problem
by Ronald Ramos 06 May '04

06 May '04

Hi All, I'm having a problem with mysql accounting, I tested a call to my mobile, after 20 secs I ended the call. But when I looked at mysql, there was no BYE, but I'm pretty sure that I got a BYE because I was looking at "ngrep 5060" from start up to end of the call. ------------------------------------------------------------------------ | sip_method | time | +----------------------------------+ | ACK | 2004-05-06 16:40:51 | | INVITE | 2004-05-06 16:40:51 | +----------------------------------+ What could be wrong? Why is it that the BYE was not inserted to the database? TIA Ronald

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sr-users May 2004