[OpenSIPS-Users] drouting 1.6.3 fails

2010-12-09 Thread Anton Zagorskiy
Hello. The drouting module trying to read 'attrs' column from a dr_rule table. There is no such column in that table. (http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id250026) Mysql log: Query select ruleid,groupid,prefix,timerec,priority,routeid,gwlist,attrs from dr_rules

Re: [OpenSIPS-Users] 477 send failed with TCP

2010-12-09 Thread Bogdan-Andrei Iancu
Hi Anshuman, do fix_nated_register() at registration time (before save(location)) if uac_nat_test() function returns true (if nat was detected) - see the nathelper module for more details: http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html Regards, Bogdan Anshuman S. Rawat

Re: [OpenSIPS-Users] drouting 1.6.3 fails

2010-12-09 Thread Bogdan-Andrei Iancu
Hi Anton, it seams your table is no longer matching the drouting version you are running. See : http://www.opensips.org/html/docs/db/db-schema-1.6.x.html#GEN-DB-DR-RULES Regards, Bogdan Anton Zagorskiy wrote: Hello. The drouting module trying to read 'attrs' column from a dr_rule

Re: [OpenSIPS-Users] Serialize Branches for same q-value

2010-12-09 Thread Bogdan-Andrei Iancu
Hi Brett, Brett Nemeroff wrote: Hello All, I'm using someone elses redirect server which is responding with several choices. None of which have a listed q-value. Now I'm told that I'm supposed to try these in order and one at a time. It seems, however, because the q-value is the same

Re: [OpenSIPS-Users] drouting 1.6.3 fails

2010-12-09 Thread Anton Zagorskiy
I'm using drouting 1.6.3 Module's documentation say nothing about 'attrs' in the dr_rules table. (See http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id250026 ) Is it possible to re-create drouting's tables? WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812

Re: [OpenSIPS-Users] OpenSIPS 1.5.3, Load balancer module and open transactions

2010-12-09 Thread Bogdan-Andrei Iancu
Hi Gavin, I just some small extension on 1.6 branch to print more info for the dialogs. So, right now, with dlg_list_ctx function, you can see the profiles the dialog belongs to. The profiles used by LB can be easily spotted by the lbX prefix in name. So list all dialogs with context and

Re: [OpenSIPS-Users] drouting 1.6.3 fails

2010-12-09 Thread Bogdan-Andrei Iancu
Ok, that needs to be updated :) just to : alter table dr_rules add column attrs CHAR(255) DEFAULT NULL; Regards, Bogdan Anton Zagorskiy wrote: I'm using drouting 1.6.3 Module's documentation say nothing about 'attrs' in the dr_rules table. (See

[OpenSIPS-Users] $shv strange errors

2010-12-09 Thread Anton Zagorskiy
Hello. I'm using $shv variables: loadmodule cfgutils.so modparam(cfgutils, shvset, load_table_domain=i:0) modparam(cfgutils, shvset, load_table_caller_id=i:0) In the main route block there are 2 'if's: (line 388) if ($(shv(load_table_domain){s.int}) == 2) route(load_table_domain);

Re: [OpenSIPS-Users] $shv strange errors

2010-12-09 Thread Duane Larson
Maybe throw some xlog statements in there print out what the variable is to see if it is actually null. You could also print out the actual caller id from the sip message with the opensips header variable in xlog. On Dec 9, 2010 6:35 AM, Anton Zagorskiy a.zagors...@oyster-telecom.ru wrote:

Re: [OpenSIPS-Users] b2bua xml files

2010-12-09 Thread Richard Revels
Ok. Thank you for the clarification on the xml and the responses to my other questions. I have quite a few other call flows where the b2bua is going to take care of things for me but I reckon this one is going to need a different solution. I have been holding on to the MediaDispatcher for

Re: [OpenSIPS-Users] $shv strange errors

2010-12-09 Thread Anton Zagorskiy
The problem was that I defined variables as integer and assigned a string value to them later. WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru From: users-boun...@lists.opensips.org

[OpenSIPS-Users] dialog and b2b data flush to disk

2010-12-09 Thread Pete Kelly
Hi I know it is possible to configure the b2b* and dialog modules to flush data to the database in case of a failure/restart of opensips. Can anyone advise if they also do a final flush to DB if the server is instructed to shutdown? For example I would like to be able to restart opensips and not

Re: [OpenSIPS-Users] db_mysql core dump

2010-12-09 Thread Anca Vamanu
Hi Duane, Investigating the modules didn't help - everything seems right. I have attached a patch that helps getting more information when the problem happens. I have to warn you that it will abort execution when observing the problem that lead to crashes. If you have real traffic through it,

Re: [OpenSIPS-Users] Freeswitch vs Asterisk

2010-12-09 Thread paul.gor...@gmail.com
I just want to reply to mr Collins with FS: your post looks very much like advertisement, and I have seen that fs is so much better than * all over internet from people connected to fs. That is unethical to say the least. In fact we have exprerienced fs crashes with core dump at least once in

[OpenSIPS-Users] random segfaults without generating core file

2010-12-09 Thread Bobby Smith
Greetings, We're encountering an issue on 1.6 trunk where we're receiving random segmentation faults that DO NOT generate cores. I've had a couple of segfaults (which I've posted to the list) that have resulted in core dumps, but doing a dmesg: opensips[28467]: segfault at 0010 rip

Re: [OpenSIPS-Users] random segfaults without generating core file

2010-12-09 Thread Duane Larson
And you're sure you have enabled Core dumps in the /etc/init.d/opensips file? if test $DUMP_CORE = yes ; then # set proper ulimit ulimit -c unlimited # directory for the core dump files COREDIR=/home/opensips/corefiles [ -d $COREDIR ] || mkdir $COREDIR chmod 777 $COREDIR

Re: [OpenSIPS-Users] Getting a Cisco 7960 to register behind a PIX

2010-12-09 Thread James Lamanna
Here's the SIP traffic from my phone now running v8.9 with nat_enable = 1 and nat_received_processing = 1. BTW this phone has no issues registering to asterisk on a different line key. -- James U nat.ip:6212 - opensips.ip:5060 REGISTER sip:opensips.ip SIP/2.0..Via: SIP/2.0/UDP

Re: [OpenSIPS-Users] Getting a Cisco 7960 to register behind a PIX

2010-12-09 Thread James Lamanna
So here's something I noticed, I'm using nat_uac_test(3) in my configuration. If you look at the REGISTER message, this test does not pass, because the NATed IP is in the Contact Header and the VIA tag. However, test 16 looks at the source port != VIA port, which would pass. I wonder if this would

Re: [OpenSIPS-Users] dialog and b2b data flush to disk

2010-12-09 Thread Anca Vamanu
Hi Pete, On 12/09/2010 04:34 PM, Pete Kelly wrote: Hi I know it is possible to configure the b2b* and dialog modules to flush data to the database in case of a failure/restart of opensips. They do that by default. The storage mechanism updates on a timer and at shutdown. Regards, --

Re: [OpenSIPS-Users] minisip with opensips over TLS not registering

2010-12-09 Thread Anca Vamanu
Hi Sanaullah, On 12/09/2010 08:26 AM, sanaullah wrote: Hi, I am trying to connect minisip client with opensips over TLS. but its can't.Error log is look like something. I am using opensips 1.6.3- with TLS ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no corresponding listening

Re: [OpenSIPS-Users] Freeswitch vs Asterisk

2010-12-09 Thread Michael Collins
Paul, It sounds like you've done a fair amount of dev work with both projects. I respect your viewpoint and your experience. Perhaps it's best if we agree to disagree. In any case, I refer back to the mantra of the FreeSWITCH developers: Use what works for your situation. If you've got something

Re: [OpenSIPS-Users] db_mysql core dump

2010-12-09 Thread Duane Larson
Anca, It failed quickly. But I am not able to do the Prints like I did last time because _h doesn't exist (gdb) backtrace #0 0x7ff75074f165 in raise () from /lib/libc.so.6 #1 0x7ff750751f70 in abort () from /lib/libc.so.6 #2 0x7ff74cc69ac1 in msg_watchers_clean (ticks=value

[OpenSIPS-Users] Handing BYE instead of CANCEL before Answer

2010-12-09 Thread Russell Bierschbach
I'm having difficulty handling a BYE message when you would normally expect a CANCEL message. According to the RFC a BYE is valid at this point... -INVITE -TRYING -183 Progress -BYE My config is here: http://pastebin.com/8Lz696wB Lines 77 to 107 are the lines in question, specifically the