sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

RE: [Users] TLS
by Helge Waastad 03 Oct '05

03 Oct '05

Hi again, I've attached the core dump. Br hw -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: 29. september 2005 19:37 To: Helge Waastad Cc: users(a)openser.org Subject: Re: [Users] TLS Hi Helge, looks like a double free - since the mem functions used by TLS are embedded, it quite hard to trace it. BTW, do you have the backtrace of the core? it will help until I'm able to reproduce the crash. regards, bogdan Helge Waastad wrote: >Hi, >sometimes when I retstart OpenSER eith tls support, I get: > >BUG: qm_free: freeing already freed pointer, first free: tls/tls_init.c: >ser_free(207) - aborting > > >br hw > > >

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[Users] outgoing packet
by ranveer kunal 03 Oct '05

03 Oct '05

Hi, I have a small problem : I am running my openser on a system(linux/freebsd) with a loopback interface(non routable) with some ip (10.102.4.139), now the incoming packet come to openser with this IP (forwarded in l2 mode), it excepts the packet and sends the reply, but the srcip of replied packet is loopback ip, instead of the routable ip of the system. Can it be configured to send the packet out using the routable ip. Regards, Ranveer.

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[Serusers] Error: I won't send a reply for ACK
by Fernando Patzlaff 03 Oct '05

03 Oct '05

Hi all, I have problem with my SER. It's work fine on the version 0.8.14. I upgrade its to 0.9.4, but when I make a call from a gateway to another gateway bypassing the SER the call is ended in 15 seconds. The message appears on tethreal is: [root@sv02 ser]# tethereal -R sip Capturing on eth0 11.426429 10.128.16.3 -> 10.128.0.11 SIP/SDP Request: INVITE sip:3354@10.128.0.11:5070, with session description 11.428174 10.128.0.11 -> 10.128.16.3 SIP Status: 401 Unauthorized 11.505784 10.128.16.3 -> 10.128.0.11 SIP Request: ACK sip:3354@10.128.0.11:5070 11.528044 10.128.16.3 -> 10.128.0.11 SIP/SDP Request: INVITE sip:3354@10.128.0.11:5070, with session description 0(3732) -> encaminhando para o gv01 asdasdasd 11.531864 10.128.0.11 -> 10.128.16.3 SIP Status: 100 trying -- your call is important to us 11.532865 10.128.0.11 -> 10.128.0.2 SIP/SDP Request: INVITE sip:3354@10.128.0.11:5070, with session description 11.591058 10.128.0.2 -> 10.128.0.11 SIP Status: 100 Trying 13.129865 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 183 Session Progress, with session description 13.130850 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 183 Session Progress, with session description 13.242219 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 180 Ringing, with session description 13.242898 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 180 Ringing, with session description 13.257439 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 13.258149 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 0(3732) ACC: transaction answered: method=INVITE, i-uri=sip:3354@10.128.0.11:5070, o-uri=sip:3354@10.128.0.11:5070, call_id=call-805E8372-0630-DA11-0214-E(a)10.128.16.3, from=<sip:10.128.16.3>;tag=a801003-12, code=200 13.406006 10.128.16.3 -> 10.128.0.11 SIP Request: ACK sip:10.128.0.11:5070;ftag=a801003-12;lr=on 0(3732) ACC: request acknowledged: method=ACK, i-uri=sip:10.128.0.11:5070;ftag=a801003-12;lr=on, o-uri=sip:10.128.0.11:5070;ftag=a801003-12;lr=on, call_id=call-805E8372-0630-DA11-0214-E(a)10.128.16.3, from=<sip:10.128.16.3>;tag=a801003-12, code=200 0(3732) Warning: sl_send_reply: I won't send a reply for ACK!! 13.755012 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 13.756575 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 14.755003 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 14.756304 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 16.755340 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 16.756279 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 45.262004 10.128.0.2 -> 10.128.0.11 SIP Request: BYE sip:10.128.0.11:5070;ftag=a801003-12;lr=on 45.262719 10.128.0.11 -> 10.128.16.3 SIP Request: BYE sip:10.128.16.3 45.336383 10.128.16.3 -> 10.128.0.11 SIP Status: 200 OK 45.336871 10.128.0.11 -> 10.128.0.2 SIP Status: 200 OK Do they have an idea? Thanks, Fernando Patzlaff

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[Serusers] preload_udomain Error
by Ryan Pagquil 03 Oct '05

03 Oct '05

Hi guys, I got these errors whenever I restarted SER. What is the meaning of these? 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 123 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 108 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 100 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 130 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 104 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 124 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 118 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 128 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 112 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 111 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 110 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 109 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 132 Could this be the problem with my aliases that are not been used by SER? 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

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FW: [Serusers] ser and * voicemail
by Rick Thompson 30 Sep '05

30 Sep '05

I have tethereal on my ser server but didn't see a 408. _____ From: Leon Sun [mailto:leon.sun@keywestcommunications.com] Sent: Friday, September 30, 2005 5:41 PM To: 'Leon Sun'; rthompson(a)vir2com.com Cc: 'Mark Aiken' Subject: RE: [Serusers] ser and * voicemail Hi, guys, I used Ethereal to trace SIP and found code 408 when it's time out. I believe we need focus 408 rather than 487. Sorry Rick, for last e-mail. Regards Leon _____ From: Leon Sun [mailto:leon.sun@keywestcommunications.com] Sent: Friday, September 30, 2005 2:33 PM To: 'rthompson(a)vir2com.com' Cc: 'Mark Aiken' Subject: RE: [Serusers] ser and * voicemail >From Mark: SER normally sends a 487 when the INVITE timer runs out so you would need to trigger the voicemail on that event. We have our feature server handing timeouts not SER (our SER timeout is set to a very large value) so I'm not sure the best way to proceed. Does SER call the failure_route for a locally generated 487 timeout? If so then rather than the 'break' you have there now just forward to vm. I would set a different t_on_failure instead of reusing "1" though, so you dont keep forwarding if the vm fails. Mark _____ From: Rick Thompson [mailto:rthompson@vir2com.com] Sent: Friday, September 30, 2005 2:27 PM To: 'Leon Sun' Subject: RE: [Serusers] ser and * voicemail Leon Thanks for the reply to my post I see what you are doing here, but that part has been taken care of. When I get a failure on "INVITE" of 486, 404, 408 or 480 I look in the mysql database and find the voicemail box for that uri and replace the sip contact then prefix it with a "V" and relay it to asterisk. Over on the asterisk side, I route the call to the voicemail context and delete the "V" and send it to voicemail. All this works except when ser times out on the "INVITE" and for some reason it doesn't route to asterisk. That's the problem I'm having. Rick _____ From: Leon Sun [mailto:leon.sun@keywestcommunications.com] Sent: Friday, September 30, 2005 4:36 PM To: rthompson(a)vir2com.com; serusers(a)lists.iptel.org Subject: RE: [Serusers] ser and * voicemail Rick, I had same problem before and I gave it up since I didn't get any answer from list. I am using another way(tricky but working) to do voice mail. Hope it can help you if you can't fix it. 1. Check location in your routing parts before relay, if not, forward to Asterisk. 2. set up unconditional forward in ATA as 00 + ATA'DID. Make a route in SER and point 00* to Asterisk. 3. Strip 00 in Asterisk and send it to voicemail2(${EXTEN}) Regards Leon Sun _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Rick Thompson Sent: Friday, September 30, 2005 12:58 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser and * voicemail Hi All I'm working with a ser script, written by someone that's gone now, that routes inbound calls to an asterisk server for voicemail. The failure_route[1], sends calls to asterisk and the IVR plays if the ua is unreachable (not in location) "404", "408" or the ua is busy "486" but it doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The call just stops ringing and 10 sec later gets a fast busy. Any ideas from anyone would be greatly appreciated. Here is the code I'm working with. failure_route[1] { xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n"); if(t_check_status("487")) { break; }; if(method=="INVITE" && (t_check_status("486|408|404|480"))) { if(avp_db_load("$ruri", "s:mailbox")) avp_pushto("$ruri/username", "s:mailbox"); prefix("V"); rewritehostport("A.B.C.D:5060"); append_branch(); xlog("L_ALERT", "****** Transfering to Voicemail\n"); t_on_reply("1"); t_relay(); }; } Thanks Rick

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[Serusers] SER Operator Pack lisence?
by Roy Sigurd Karlsbakk 30 Sep '05

30 Sep '05

hi i just came by http://www.iptelorg.de/SOP.php, and it's quite clearly SER, but then, on the downloads site, it tells me "This site offers only commercial products and services . Please follow the links below if you wish to find support related to respective software". although their stuff looks good, can you really make GPLed code a "commercial product"? roy

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FW: [Serusers] ser and * voicemail
by Rick Thompson 30 Sep '05

30 Sep '05

Mark Thanks for the re-post What I see in ngrep and var/log/messages is a "CANCEL" from the ua and a failure "487" on a reply. Don't know which happens first. Even if I take out the "break" if and add "487" to the if(method== statement, it goes to the asterisk server but vm doesn't play, no audio, and it drops the call anyway. I think I'll try what you suggested about creating another t_on_failure[] and test for it within the main routing block. Rick _____ From: Mark Aiken [mailto:aiken.mark@gmail.com] Sent: Friday, September 30, 2005 5:22 PM To: Leon Sun Cc: rthompson(a)vir2com.com; serusers(a)lists.iptel.org Subject: Re: [Serusers] ser and * voicemail SER normally sends a 487 when the INVITE timer runs out so you would need to trigger the voicemail on that event. We have our feature server handing timeouts not SER (our SER timeout is set to a very large value) so I'm not sure the best way to proceed. Does SER call the failure_route for a locally generated 487 timeout? If so then rather than the 'break' you have there now just forward to vm. I would set a different t_on_failure instead of reusing "1" though, so you dont keep forwarding if the vm fails. Mark On 9/30/05, Leon Sun <leon.sun(a)keywestcommunications.com> wrote: Rick, I had same problem before and I gave it up since I didn't get any answer from list. I am using another way(tricky but working) to do voice mail. Hope it can help you if you can't fix it. 1. Check location in your routing parts before relay, if not, forward to Asterisk. 2. set up unconditional forward in ATA as 00 + ATA'DID. Make a route in SER and point 00* to Asterisk. 3. Strip 00 in Asterisk and send it to voicemail2(${EXTEN}) Regards Leon Sun _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org <mailto:nces@iptel.org> ] On Behalf Of Rick Thompson Sent: Friday, September 30, 2005 12:58 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser and * voicemail Hi All I'm working with a ser script, written by someone that's gone now, that routes inbound calls to an asterisk server for voicemail. The failure_route[1], sends calls to asterisk and the IVR plays if the ua is unreachable (not in location) "404", "408" or the ua is busy "486" but it doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The call just stops ringing and 10 sec later gets a fast busy. Any ideas from anyone would be greatly appreciated. Here is the code I'm working with. failure_route[1] { xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n"); if(t_check_status("487")) { break; }; if(method=="INVITE" && (t_check_status("486|408|404|480"))) { if(avp_db_load("$ruri", "s:mailbox")) avp_pushto("$ruri/username", "s:mailbox"); prefix("V"); rewritehostport("A.B.C.D:5060"); append_branch(); xlog("L_ALERT", "****** Transfering to Voicemail\n"); t_on_reply("1"); t_relay(); }; } Thanks Rick _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] ser and * voicemail
by Rick Thompson 30 Sep '05

30 Sep '05

Hi All I'm working with a ser script, written by someone that's gone now, that routes inbound calls to an asterisk server for voicemail. The failure_route[1], sends calls to asterisk and the IVR plays if the ua is unreachable (not in location) "404", "408" or the ua is busy "486" but it doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The call just stops ringing and 10 sec later gets a fast busy. Any ideas from anyone would be greatly appreciated. Here is the code I'm working with. failure_route[1] { xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n"); if(t_check_status("487")) { break; }; if(method=="INVITE" && (t_check_status("486|408|404|480"))) { if(avp_db_load("$ruri", "s:mailbox")) avp_pushto("$ruri/username", "s:mailbox"); prefix("V"); rewritehostport("A.B.C.D:5060"); append_branch(); xlog("L_ALERT", "****** Transfering to Voicemail\n"); t_on_reply("1"); t_relay(); }; } Thanks Rick

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[Serusers] loose routing of BYE from UDP to TCP
by Mark Aiken 30 Sep '05

30 Sep '05

Hi, I'm having problems getting SER to (loose) route requests from UDP to TCP. I'm not sure if the UA is at fault here or some SER config issue. SER receives an INVITE over TCP, record_route() is called and then the request is relayed via UDP (t_relay_to_udp) to the UA. The INVITE relayed to the UDP UA now has 2 Record-Route headers added by SER. One has the transport=tcp parameter. Record-Route: <sip:xx.xx.xx.xx;r2=on;ftag=xyz;lr=on> Record-Route: <sip:xx.xx.xx.xx;transport=tcp;r2=on;ftag=xyz> When the UA sends the BYE to SER, it has the 2 Route headers like so (on a single line): Route: <sip:xx.xx.xx.xx;r2=on;ftag=xyz;lr=on>,<sip:xx.xx.xx.xx ;transport=tcp;r2=on;ftag=xyz;lr=on> The SER script simply calls t_relay() in the loose_route section of the script. I expected t_relay() to relay the BYE via TCP, since the Route header has transport=tcp, but it sends the BYE via UDP instead. Anyone seen this problem before? Do I need to check for transport=tcp and call t_relay_to_tcp, rather than using t_relay after loose_route()? Mark

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Re: [Serusers] Windows to Linux
by zkeatts 30 Sep '05

30 Sep '05

I am probably not the best person to ask that, since I have been brute forcing my way through - i.e. reading and debugging code without consulting a manual. There are tons of resources on the net for coding in C and since you already have 3 years of experience I am sure that moving to ANSI C won't be too hard. If you have coded in JAVA or C++ then it should be a breeze. If anything you should read up on pointers and memory allocation, because those two seem to catch most new coders the most. The GDB suggestion is also excellent. I have found it to be one of the most useful debugging tools that I have ever used since it gives a nice stack backtrace with line numbers etc. good luck, Zach Keatts Software Engineer Nuvio.com viquar syed wrote: > Hey Zach, > First off, thanks for your reply. Yours is the first one I got beside > the OUT OF OFFICE notices. > I plan on getting the installation up. I want to get down to Stack > level, but I guess learning ANSI C would be my first step, have your > foud anything useful towards this ? > Again I appreciate your input. > > Thanks > Viq > DBA/Developer > Cordia Communications > Orlando FL > 407-313-7015 > > On 9/30/05, *zkeatts* <zkeatts(a)nuvio.com <mailto:zkeatts@nuvio.com>> > wrote: > > I am in a similar situation, as far as the ANSI C goes. What has > helped > me is to jump in, find a bug and try to figure out what is going on. > Granted you might not be as lucky as me on your first try in finding > bugs, but trust me even if you don't there is plenty to look > at. I can > promise you it will be confusing and infuriating, but slowly > things will > begin to make sense. > > Start by setting up a basic SER installation on a Linux box. Once you > get it set up and working then you can try out different features. > Don't be afraid to send messages to this group, I have found them > very > helpful and attentive to all levels of questions. > > good luck, > > Zach Keatts > Software Engineer > Nuvio.com <http://Nuvio.com> > > viquar syed wrote: > > > Hi, > > My company wants me to hop on to Linux/Unix from Bill Gates, and to > > make it even better I am suppose to get upto speed writing > modules to > > enhance our SER deployment. > > What experience do I have writing in ANSI C "0", "ZERO", "nada", > > BUT, I have been programming for over three yrs now so the logic is > > not an issue. > > My question, whats the best way to start, I am currently going > through > > the documentation on SER Admin, need to have a plan. > > I know no one likes newbees, but hey any help would be highly > appreciated. > > Thanks > > Viq > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

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sr-users September 2005