Re: [OpenSIPS-Users] Cancel branch before failover on timeout

bog...@opensips.orgwrote: ** Hi Ronald, I wouldn't go so far - even if you get 2 records for the transaction based accounting, the values will be mixed. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 07/10/2013 03:08 PM, Ronald Cepres

Re: [OpenSIPS-Users] Cancel branch before failover on timeout

Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 07/04/2013 07:41 PM, Ronald Cepres wrote: Bogdan, Thanks for the informative reply. What I really want to solve is a problem I encounter when the first GW doesnt respond after a defined timeout

Re: [OpenSIPS-Users] Cancel branch before failover on timeout

branches failed. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 07/03/2013 10:43 PM, Ronald Cepres wrote: Hi all, Is there a way I can cancel a pending branch before doing a fail-over to next gateway (due to timeout from previous gateway

[OpenSIPS-Users] Cancel branch before failover on timeout

Hi all, Is there a way I can cancel a pending branch before doing a fail-over to next gateway (due to timeout from previous gateway)? This way I can make sure that the call to the previous gateway will not go through anymore after fail-over to the next gateway, thus preventing us double-charged

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [DROUTING] Crash on use_next_gw/get_gw_by_id

is available for testing (see attached). Please have it tested and let me know if it fixes the problem. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 05/16/2013 07:28 PM, Ronald Cepres wrote: Drouting crashes when selecting next gateway. Did

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [DROUTING] Crash on use_next_gw/get_gw_by_id

On 05/20/2013 07:40 PM, Ronald Cepres wrote: Hi Bogdan, Thanks for the reply. However, the patch that is attached is empty (0 bytes and doesn't contain anything). Regards, On Tue, May 21, 2013 at 12:36 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hello, Thanks to the help

Re: [OpenSIPS-Users] is_method crash when used on pike triggered route

Hi Bogdan, OpenSIPS didn't crash anymore after I used your patch. Thanks for the help! Regards, Ronald On Wed, Jan 18, 2012 at 11:35 PM, Ronald Cepres rbcep...@gmail.com wrote: Hi Bogdan, Thanks for the quick patch. I'll try it out tomorrow and let you know the result at the end of the day

Re: [OpenSIPS-Users] is_method crash when used on pike triggered route

, Bogdan On 01/16/2012 06:08 AM, Ronald Cepres wrote: Hi Bogdan, Thanks for your reply. The crash happened often (every less than hour) on live traffic, but I was not able to reproduce the bug on my own. Here's the last part of OpenSIPS logs that I saved after the crash: Jan 13 09:31

[OpenSIPS-Users] is_method crash when used on pike triggered route

Hi all, I'm using OpenSIPS 1.7.1 and based from the attached back trace, it crashed when it is trying to parse the method of a REGISTER message received by the server, triggered by pike route. Here is a snippet of my opensips.cfg: ... loadmodule pike.so modparam(pike, sampling_time_unit, 30)

[OpenSIPS-Users] Source IP address on Asterisk integration

Hi all, I'm trying to set up Opensips so that it simply relays the requests it receives to Asterisk on the same server, only using a different port. The set-up is working but my problem is Asterisk uses the IP of OpenSIPS as peer contact even if the domain on the Contact header is from the actual

Re: [OpenSIPS-Users] Opensips crashing due to out of memory error

Have you tried increasing the amount of allocated memory for OpenSIPS? On Sun, Apr 10, 2011 at 5:53 AM, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote: Hi, I'm getting random crashes of Opensips 1.6.2, here are the entries in the log, seems to be out of memory, how should i try to solve

Re: [OpenSIPS-Users] sst module killing calls

I also tried parsing the Session-Expires header value and re-set the timeout_avp to this value during re-INVITE. The results are still the same. Has anyone encountered this problem and fixed it? Any help would be appreciated. Thanks! Regards, Ronald On Thu, Mar 3, 2011 at 2:58 AM, Ronald Cepres

Re: [OpenSIPS-Users] ratelimit: per group/account limiting

95 every second) and all calls after that will be rejected. Hope this helps. Regards, Ovidiu Sas On Wed, Feb 23, 2011 at 12:51 PM, Ronald Cepres rbcep...@gmail.com wrote: On Wed, Feb 23, 2011 at 6:10 AM, Ovidiu Sas o...@voipembedded.com wrote: If a virtual PRI is set up (23

Re: [OpenSIPS-Users] How to use sst to account missing BYEs

be wrong with my setup? Regards, Ronald On Mon, Jan 10, 2011 at 8:02 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Ronald, If you look at the SIP capture, do you see the Session-Timer header sent to callee ? does your caller/callee supports SST ? Regards, Bogdan Ronald Cepres

Re: [OpenSIPS-Users] sst module killing calls

Hi Bogdan, I get a very similar behavior as what Jeff gets. I can see the re-INVITE coming and I'm sure that it goes through loose_route block. I can also see the Update by a REQUEST log. Here's a snippet of the logs after getting the re-INVITE: DBG:dialog:dlg_onroute: route param is

Re: [OpenSIPS-Users] ratelimit: per group/account limiting

at 12:51 PM, Ronald Cepres rbcep...@gmail.com wrote: On Wed, Feb 23, 2011 at 6:10 AM, Ovidiu Sas o...@voipembedded.com wrote: If a virtual PRI is set up (23 channels for NA or 30 channels for Europe), again the cps doesn't really count. As soon as the virtual PRI is maxed out

Re: [OpenSIPS-Users] pseudo-variable problem with increasing cps

: if ( t_local_replied(all) ) { xlog(did not get any response); } else { xlog($(replyci): $C(rx)failure route: $(replyrs) $(replyrr)$C(xx)\n); } Dave On Wed, Feb 23, 2011 at 10:04 AM, Ronald Cepres rbcep...@gmail.comwrote: Hi all, I'm setting up opensips as a stateful proxy, and i have

Re: [OpenSIPS-Users] pseudo-variable problem with increasing cps

not get any response); } else { xlog($(replyci): $C(rx)failure route: $(replyrs) $(replyrr)$C(xx)\n); } Dave On Wed, Feb 23, 2011 at 10:04 AM, Ronald Cepres rbcep...@gmail.comwrote: Hi all, I'm setting up opensips as a stateful proxy, and i have the following snippet of code

Re: [OpenSIPS-Users] ratelimit: per group/account limiting

On Wed, Feb 23, 2011 at 6:10 AM, Ovidiu Sas o...@voipembedded.com wrote: If a virtual PRI is set up (23 channels for NA or 30 channels for Europe), again the cps doesn't really count. As soon as the virtual PRI is maxed out (in terms of channels) all subsequent calls will be rejected (and

[OpenSIPS-Users] pseudo-variable problem with increasing cps

Hi all, I'm setting up opensips as a stateful proxy, and i have the following snippet of code on the failure route: failure_route[1] { xlog($(replyci): $C(rx)failure route: $(replyrs) $(replyrr)$C(xx)\n); # Failure route routine... } The values of the call-id, response code and reason

[OpenSIPS-Users] ratelimit: per group/account limiting

Hi everyone, I would like to ask all of you if it is possible to use ratelimit module to limit cps per account/group (i.e.: account A has limit of 10 cps, account B's is 20 cps, etc.)? Is it even possible to implement this set-up on opensips? Thanks for any kind of help. Regards, Ronald

[OpenSIPS-Users] Weird routing of ACK on INVITE

Hi all, After relaying an INVITE from my PBX to the carrier side (and after carrier responded as expected), I receive an expected ACK from my PBX with the domain part of the RURI of the ACK is the same as the IP of my OpenSIPS, which seems to be impossible to route back to the carrier side

[OpenSIPS-Users] dialog flag and create_dialog function

Hi to all, Does setting the dialog flag and calling the create_dialog function create redundant dialogs for a transaction? Just wondering since I didn't find it indicated in the dialog module documentation. Thanks! Regards, Ronald ___ Users mailing

Re: [OpenSIPS-Users] dialog flag and create_dialog function

Ronald Cepres wrote: Hi to all, Does setting the dialog flag and calling the create_dialog function create redundant dialogs for a transaction? Just wondering since I didn't find it indicated in the dialog module documentation. Thanks! Regards, Ronald

Re: [OpenSIPS-Users] drouting: no valid routing rules

. - Original Message - *From:* Ronald Cepres rbcep...@gmail.com *To:* OpenSIPS users mailling list users@lists.opensips.org *Sent:* Friday, January 07, 2011 19:16 *Subject:* Re: [OpenSIPS-Users] drouting: no valid routing rules Hi Bogdan, yes it is the right url. anyway, i solved my

[OpenSIPS-Users] How to use sst to account missing BYEs

Hi to all! I've been setting up a media-less SIP proxy server using OpenSIPS. We need to account (and bill) every call that goes through the proxy and one of our main concerns is the issue of missing BYEs wherein we can't account the call without a BYE. I tried to use sst module to solve this

[OpenSIPS-Users] drouting: no valid routing rules

Hi to all! After upgrading from 1.6.3 to 1.6.4, I keep on getting this log when starting OpenSIPS: WARNING:drouting:dr_load_routing_info: table dr_rules is empty DBG:drouting:dr_load_routing_info: 0 records found in dr_rules WARNING:drouting:dr_load_routing_info: no valid routing rules -

Re: [OpenSIPS-Users] drouting: no valid routing rules

to the right database ? Regards, Bogdan Ronald Cepres wrote: Hi to all! After upgrading from 1.6.3 to 1.6.4, I keep on getting this log when starting OpenSIPS: WARNING:drouting:dr_load_routing_info: table dr_rules is empty DBG:drouting:dr_load_routing_info: 0 records found in dr_rules

Re: [OpenSIPS-Users] Drouting module parameters not found

Thanks Bogdan! Regards, Ronald On Mon, Dec 20, 2010 at 10:48 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Ronald, use the SVN check out for 1.6 branch - http://www.opensips.org/Resources/Downloads#svn Regards, Bogdan Ronald Cepres wrote: Not actually from SVN but from

Re: [OpenSIPS-Users] Not enough free memory

/opensips.html Thanks! Regards, Ronald On Fri, Dec 17, 2010 at 1:56 AM, Ovidiu Sas o...@voipembedded.com wrote: Here's how to investigate/debug memory issues: http://www.opensips.org/Resources/DocsTsMem Regards, Ovidiu Sas On Thu, Dec 16, 2010 at 12:44 PM, Ronald Cepres rbcep...@gmail.com

[OpenSIPS-Users] Not enough free memory

Hi to all, If I run OpenSIPS (1.6.3 ) for a long time while calls are coming in, it suddenly stops with the following sample logs: Dec 16 12:26:43 [12114] ERROR:core:new_avp: no more shm mem Dec 16 12:26:43 [12114] ERROR:core:add_avp: Failed to create new avp structure Dec 16 12:26:43 [12114]

Re: [OpenSIPS-Users] Drouting module parameters not found

the same. On Mon, Dec 13, 2010 at 9:25 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Ronald, Are you sure you have the latest 1.6.3 sources from SVN ? Regards, Bogdan Ronald Cepres wrote: Hi to all, I have a problem about the drouting module. Here is a snippet of my script

[OpenSIPS-Users] Drouting module parameters not found

Hi to all, I have a problem about the drouting module. Here is a snippet of my script configuration: ... loadmodule drouting.so modparam(drouting, db_url, mysql://user:p...@localhost/opensips) modparam(drouting, gw_id_avp, $avp(s:gw_id)) modparam(drouting, rule_id_avp, $avp(s:rule_id)) ...