sr-users December 2007

sr-users@lists.kamailio.org

130 participants
234 discussions

[Serusers] ser-0.9.6: modules/postgres bugreport & patch

by Alexandre Snarskii

Hi! I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend is postgresql version 8.1.3. Today I got errors in logfiles, saying: messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]: ERROR:avpops:dbrow2avp: dbrow contains NULL fields The similar problem reported in: http://lists.iptel.org/pipermail/serusers/2005-May/019681.html with much more detailed description of error, database contents and config samples. Patch is trivial, and looks more like a fix to copy'n'paste error: in mysql/val.c function str2val states: if (!_s) { memset(_v, 0, sizeof(db_val_t)); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } VAL_NULL(_v) = 0; and the last line mentions that 'well, that's value is not NULL'. In postgresql/db_val.c, line 182, function str2valp, the same statement is the: if (!_s) { DLOG("str2valp", "got a null value"); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } without explicit notification that this is not-NULL value. More than, nowhere else in this function VAL_NULL(_v) not set to 0. So, if a value _v.val contained anyting but 0 at the function start, resulting value will be threated as NULL despite the fact, that _s is NOT NULL. Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and everyting will be ok. At least for me it's ok for some hours :)

14 years, 5 months

[Serusers] SER Nokia CANCEL Problem

by Abdul Qadir

Hi , I tried to call from one nokia sip (E61 and other models )phone to another nokia sip phone. The call works fine. The problem comes only when I call from Phone A to Phone B and then immediately cancel the call(from Phone A). The Phone A will hangup the call as it sent CANCEL but the SER will ignore this CANCEL and still send INVITE to Phone B resulting in a ghost call situation. I tried to capture a log of message and found that Phone A "CANCEL" message is received on SER even before any provisional response from Phone B. Therefor SER doesnot relay this CANCEL request to Phone B. I even checked RFC which clearly says that UAC should not send CANCEL untill it receives any provisional response. I talked to Nokia expert and they said the 100 Trying message from your server is considered as provisional response, therefor behaviour of client is absolutely correct. Is there any way I can stop 100 Trying message and still run statefull SER, so that I can verify what nokia said. Any ideas suggestions are welcome. Thanking you all in advance. Best Regards, Abdul Qadir --------------------------------- Don't be flakey. Get Yahoo! Mail for Mobile and always stay connected to friends.

14 years, 10 months

[Serusers] Fwd: SIP Express Bundle ready-to-go idea, request for input

by Mike Trest - Personal

Greger, et. al., I like your idea enough to volunteer as a tester + documentation writer. I do a lot of tech and user writing. Active in VoIP since the beginning. I am active VoIP engineer and developer. Also PSTN & VoIP network designer-builder. I have my own private machines, network, and VoIP gateway resources. Prefer CentOS or FC* ..mike.. >Hi guys, >I have been playing with the following idea: >Create a ready-to-run OS image with everything that is needed for >iptel.org apps pre-installed + a complete installation of: >* SER 2.0 (release) >* rtpproxy >* SEMS >* SERweb >* maybe sipsak, some monitoring tools, etc > > >The idea is to create a small script 'config_iptelorg' for >configuring the installation to your needs. You should then be able >to download the ready image, boot it, go through the script and have >an up and running iptel.org proxy and app server just like the >iptel.org free SIP service in maybe 10-15 minutes. This way you >could host a SIP service for your own domain with close to no setup at all. > >Some questions to you: >* Is there any interest for this at all? >* I was thinking about using Ubuntu 7.10 server as the OS. Any >thoughts/preferences? >* Should the image be an Amazon EC3 image (you could use >http://www.rightscale.com and get it running in no time with 10 >run-hours free) or should it be a VMware appliance to be run with >free VMware Player? >* Other suggestions/comments? > >I could need some help with this, anyone interested in lending me a >hand? (could be anything, documentation, testing, installation, etc) >g-)

15 years

[OpenSER-Users] strange error logs in syslog

by Cseke Tamas

Hello, We have a lots of error log like this, what fills syslog ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction ERROR:tm:w_t_relay: t_forward_nonack failed ERROR: t_check_status: cannot check status for a reply which has no T-state established Could you tell me how could i avoid this? I think these are not errors. Shall i just change the debug level for this logs? Thanks any help, Tamas

15 years, 2 months

[OpenSER-Users] Fix up source address.

by Alex Balashov

Greetings, I have a strange problem using OpenSER 1.3.x with nathelper. Two ethernet interfaces: eth0 = 192.168.0.0/24 eth1 = outside.ip/29 For some reason, no matter what I do to mangle the requests with nathelper's functions, the packet is *always* sent out of eth1 with the *source address* of the machine's eth0. Obviously, the response from the far-end SIP peer never gets back. The packet does physically go over eth1, I know that much from packet captures. I don't even see how this is possible; when OpenSER issues a packet, shouldn't it originate according to the machine's routing table, take the most specific route, and consequently, adopt the right source address? What gives? I have OpenSER 'listening' on both interfaces, and have tried both on and off with this setting. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

15 years, 2 months

[Serusers] SER 2_0_0 ERROR: slow timer too slow

by tzieleniewski

Hi, I encountered the following error: Jun 15 08:57:21 rd ser[12680]: ERROR: slow timer too slow: overflow (7997 - 6974 = 1023) Jun 15 08:57:24 rd last message repeated 46 times Could you please point me what could be the reason? Thanks in advance. Tomasz

15 years, 2 months

[OpenSER-Users] INVITE, exec_dset(), and BYE

by Juan Carlos Castro y Castro

I have a scenario in which an OpenSER machine distributes load among several Asterisk machines for outgoing PSTN. I use an external program which I call through exec_dset() to select which server a call goes to. I use record_route(). All calls are completed OK, but when I dial from some SIP user-agents and hang up from the caller side, OpenSER gets crazy and doesn't know where to route the BYE to. Here's the reason: Good scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:providercode#5551234@chosenserver SIP/2.0" - I get an OK back, all is well. (Sometimes the # in the uri above is sent as %23, but it works either way) Bad scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:5551234@mydomain SIP/2.0" - My openser.cfg doesn't know how to handle that and I get a "Loop Detected" back. The destination never gets a hangup signal. Is the second UA from the "bad" scenario disrespecting the RFC? Do I have any recourse to route a BYE that comes like that to the right Asterisk server? Thanks, Juan

15 years, 2 months

[OpenSER-Users] Openser + Mediaproxy behind NAT

by Halomoan Chow

Hi Is it possible to put Openser + Mediaproxy behind NAT while serving internal phones and external phones ? Internal phone <-> Openser+MediaProxy <-> NAT <-> external phone (maybe behind nat also) Openser is act as a registrar in this case. If you have successful story with this setting, please share. Thank you so much Regards, Halomoan

15 years, 2 months

[OpenSER-Users] Run on multiple ports

by Arun Kumar

Hi, how do I run ser/openser on multiple ports. ? I've configured my ser with two listen stat. listen = ip:5060 listen = ip:5070 UA:5070 -------> Cisco (Blocked 5060) --------> SER -------------> UB:5060 UA is able to call UB, but UB is not able to call UA, I know 5060 is blocked but it should goto 5070 ? please guide on this ? Thanks, Arun

15 years, 2 months

Re: [OpenSER-Users] newbie help - handeling 302s and 404s

by Jesus Rodriguez

Hi Lazer, >i need a simple config that achieves the following (w/ out a db) > >if a user is not registered then all calls should go to a default gw >192.168.10.1 > >the default gw will only respond w/ a 302 or a 404 > >if a 302 is received then openser should forward the call on to the >new gw listed in the reply > >if a 404 is received the call should be forwarded to a different gw 172.16.20.1 > >if the send gw is done then the call should die. Try with somehting like this (not tested): if (!lookup("location")) { route(5); exit; } route[5] { rewritehostport("192.168.10.1:5060"); t_on_failure("1"); route(1); exit; } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (t_check_status("404")) { rewritehostport("172.16.20.1:5060"); append_branch(); t_relay(); } else { get_redirects("*"); t_relay(); } } Saludos JesusR. >here is what i was thinking of but as a newbie i may be off my rocker >i seem to get an error > > ERROR: t_check_status: cannot check status for a reply which has no >T-state established > >but again my whole approach may be wrong. > > > if (!lookup("location")) { > >rewritehostport ("192.168.10.1:5060"); >if (t_check_status("(404)|(480)")) { > log("404 or 480 negative reply\n"); >rewritehostport ("172.16.20.1:5060"); >} >else{ >t_on_failure("1"); > t_relay(); >} > > > failure_route[1] { > > get_redirects("*"); > t_relay(); >} > } Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

15 years, 2 months

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sr-users December 2007