[OpenSIPS-Users] Does OpenSIPS support RFC 5626 (Require Outbound)

I was doing some testing with Snom phones to try and get the phone to register to two Edge Proxies at once. With Snom you are able to configure multiple outbound proxies for a user. Snom sends a REGISTER to the first proxy without issue but the 200 response back from OpenSIPS doesn't have the fol

Re: [OpenSIPS-Users] OpenSIPS resolving DNS SRV records

Thanks Adrian. I'm asking my vendor why there are no UDP SRV records. On Sun, May 24, 2015 at 7:04 AM, wrote: > Try calling @sip2sip.info, it is a domain with proper DNS records for SIP. > > Adrian > > > On 22 May 2015, at 16:27, Duane Larson wrote: > > O

Re: [OpenSIPS-Users] OpenSIPS resolving DNS SRV records

, to a plain domain > with no transport parameter or port, opensips will do a udp SRV lookup on > the domain. > > Can you provide the actual domain please? If you cant - do a srv lookup on > it to verify it actually has an SRV. > > dig _sip._udp.blabla.webex.com SRV > >

[OpenSIPS-Users] OpenSIPS resolving DNS SRV records

OpenSIPS does not appear to be resolving the SIP address of a domain that is provided by WebEx.com. When I dial dlar...@blahblah.webex.com the call gets forwarded to the IP address of the webserver but it really should be going to the IP address from an SRV record. I believe the OpenSIPS default

Re: [OpenSIPS-Users] Testing Event_Route with USRLOC

operhttp://www.opensips-solutions.com > > On 10.02.2015 06:31, Duane Larson wrote: > > I'm trying out event_route with USRLOC but I am not seeing any xlogs. I'm > not 100% clear how to set up event_route's but it seems like it is > basically just creating the follo

[OpenSIPS-Users] Testing Event_Route with USRLOC

I'm trying out event_route with USRLOC but I am not seeing any xlogs. I'm not 100% clear how to set up event_route's but it seems like it is basically just creating the following modparam("usrloc", "db_mode", 2) event_route[E_UL_AOR_INSERT] { xlog("The E_UL_AOR_INSERT event was raised\n");

Re: [OpenSIPS-Users] Multi-Proxy Environment Routing

order to sync the cache too. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 25.01.2015 02:31, Duane Larson wrote: > > Before I try to reinvent the wheel I wanted to see if there is already a > way to do this.

[OpenSIPS-Users] Multi-Proxy Environment Routing

Before I try to reinvent the wheel I wanted to see if there is already a way to do this. For redundancy I have two Proxies and I am using DNS SRV with Proxy01 weighted higher so that it is the primary that all clients register and use. Both proxies use the same location database. In the event of

Re: [OpenSIPS-Users] Block user from registration

in > Opensips so i can switch on/off user base registration? ( We only allowing > to send calls outside, no inbound calls allowed) > > Hope it helps you to understand my scenario, Let me know if i am wrong > anywhere in above scenario. > > On Wed, Dec 31, 2014 at 1:30 PM, Duane Lar

Re: [OpenSIPS-Users] Block user from registration

register if they are from a friendly IP. On Wednesday, December 31, 2014, Satish Patel wrote: > How it will help if i want to allow only IP auth for specific user but not > registration auth? How your logic deal with User level? > > > On Wed, Dec 31, 2014 at 12:22 PM, Duane Larson >

Re: [OpenSIPS-Users] Block user from registration

Would you not just do something like this? If(FriendlyIP && is_method("REGISTER")) { if (t_newtran()) { save("location"); } exit; } On Wed, Dec 31, 2014 at 10:22 AM, Satish Patel wrote: > Hi, > > We have many users using b

[OpenSIPS-Users] dlg_val() is inconsistent

I was running version 1.9 and just upgraded to 1.11.2. I was having my issue on 1.9 and thought a newer version might fix my issue but it is still occuring. I am using dlg_val to keep up with parked calls on Asterisk so that if someone needs to retrieve a parked call the caller is relayed to the

Re: [OpenSIPS-Users] Upgrading from 1.9 to 1.11 - usrloc issue

ype instead of > the old "INT" type. > > Best regards, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 18.08.2014 01:50, Duane Larson wrote: > > I am trying to upgrade to the latest version and I am running into some > segfault

[OpenSIPS-Users] Upgrading from 1.9 to 1.11 - usrloc issue

I am trying to upgrade to the latest version and I am running into some segfault issues I've update the database version numbers and don't have any WARNINGs showing up when I start OpenSIPS. Here is the issue I see in syslog Aug 17 17:44:51 SIPProxy02 kernel: [25783210.686957] opensips[18086]: s

Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

Great! Thanks On Thursday, August 7, 2014, Tijmen de Mes wrote: > Hi, > > It is in CDRTool. Soon we will release a new version of call control which > should be safe if OpenSIPS does not have support for this. > > -- > Tijmen de Mes > AG-Projects > > On 7 augustus 2014 at 04:16:01, osiris123d (d

Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 1.11 beta major release is out

Good job. Looks like the "Read More . . ." link for "New Call_Center Module" on page http://www.opensips.org/About/Version-1-11-0 is wrong It is pointed to " http://www.opensips.org/html/docs/modules/1.12.x/call_center.html"; Think it needs to be http://www.opensips.org/html/docs/modules/1.11.x/

Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

Ugh Nevermind. My "Max Duration" was set to 6 on the Destinations rate setup. I set it to zero and I think things are looking better now. Think I'm done for the day. Good work David! Really appreciate the patch. On Sat, Jan 25, 2014 at 6:56 PM, Duane Larson wrote: >

Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

e outbound calls all cost $0.0005. On Sat, Jan 25, 2014 at 6:24 PM, Duane Larson wrote: > David/Tijmen/Adrian, > > It is working for me too. Both inbound and outbound are being recognized > and the different rates are being applied. I will keep looking at it while > more calls ar

Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

rk on updating the docs. > > Thanks! > -- > David M. Lee > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > On Jan 23, 2014, at 4:43 AM, Adrian Georgescu wrote: > >

Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

vis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > On Jan 23, 2014, at 4:43 AM, Adrian Georgescu wrote: > > > I think it would be a good idea. > > > > Adrian > > > > On 22 Jan 2014, at 16:58, David Lee (di

Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

If you are going to make a patch I might be interested in testing it as long as AG-Projects are ok with adding it into future version. On Wed, Jan 22, 2014 at 12:58 PM, David Lee (digium) wrote: > Duane Larson wrote: > > I have been playing with CDRTool for a while but I am not sure

[OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

I have been playing with CDRTool for a while but I am not sure if it is possible to rate Origination (Inbound) calls differently than Termination (Outbound) calls from my SIP Provider. For Origination I pay 0.0035 and for Termination I pay 0.005. Keep in mind these costs are for destination "1".

Re: [OpenSIPS-Users] Call Pickup Feature

I think there might be a little confusion here and a language barrier. If I am understanding Jorge correctly I think he is stating that OpenSIPS is load balancing between many Asterisk servers. OpenSIPS can do the Call Pickup feature as can Asterisk. I think the issue is that if OpenSIPS is doin

Re: [OpenSIPS-Users] Where can cachedb_url be used

Thanks Laszlo On Wed, Jul 24, 2013 at 4:57 AM, Laszlo wrote: > It is not well documented, but it's known to be working with usrloc, > dialog, permissions and drouting. > You may need to try it out with other modules, it won't be hard to verify > :) > > -Laszlo

[OpenSIPS-Users] Where can cachedb_url be used

I know that with the NoSQL modules you have the "Exported Parameter" to set up cachedb_url example modparam("cachedb_mongodb","cachedb_url","mongodb:instance1://localhost:27017/db.collection") I saw in a post that it is possible to do the following with the Dialog module modparam("dialog","cach

[OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor

solved (with the REGISTERs) ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 04/29/2013 11:41 PM, Duane Larson wrote: > > Wow those thresholds give you a good amount of info. I'll have to see h

Re: [OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor

ntent-Length: 0#015#012#015#012 Apr 29 15:31:38 SIPProxy02 /usr/local/sbin/opensips[3106]: WARNING:core:log_expiry: #1 is a core action : 6 - 513533us - line 553 Apr 29 15:31:38 SIPProxy02 /usr/local/sbin/opensips[3106]: WARNING:core:log_expiry: #2 is a core action : 14 - 513520us - line 595 Apr 29 1

Re: [OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor

s/modules/1.9.x/db_mysql.html#id249058 > > http://www.opensips.org/Documentation/Script-CoreFunctions-1-9#toc51 > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 04/29/2013 11:04 PM, Duane Larson wrote: > > I w

Re: [OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor

Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 04/26/2013 03:23 AM, Duane Larson wrote: > > I originally posted this via Nabble but I am not sure if it went to the > Opensips User mailing list so please excuse me if this shows up as multiple > posts

[OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor

I originally posted this via Nabble but I am not sure if it went to the Opensips User mailing list so please excuse me if this shows up as multiple posts. I am starting to see this issue a lot lately. My Snom phones will so as not registered on their display screen and when I look in the syslogs

[OpenSIPS-Users] Presence_XML issue with version 1.9

I just installed version 1.9 and I am having an issue starting OpenSIPS. It looks like with version 1.9 you can no longer use modparam("presence_xml", "db_url" modparam("presence_xml", "integrated_xcap_server", Now that I've taken that out I get the following error when trying to start Mar 31 0

Re: [OpenSIPS-Users] Issue with From domain coming from Asterisk

evel. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 02/28/2013 06:10 PM, Duane Larson wrote: > > Yeah. I figure with the Dialog module I will need to save the from domain > before I send it to Asterisk and

Re: [OpenSIPS-Users] Issue with From domain coming from Asterisk

what Asterisk can do. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 02/28/2013 05:56 PM, Duane Larson wrote: > > I kind of figured this but just wanted to check since that post about > Asterisk and the

Re: [OpenSIPS-Users] Issue with From domain coming from Asterisk

> domain on opensips - it is not something complex to do and you can use the > dialog support for that to avoid any dependency from the end-point devices . > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 02/2

[OpenSIPS-Users] Issue with From domain coming from Asterisk

I wanted to see if I could get this answered on the OpenSIPS mailing list even though this kind of has to do with how Asterisk works. I am hoping someone has run into this and figured a way to resolve the issue. I have OpenSIPS set up to be a proxy for a cluster of Asterisk servers. When a call

Re: [OpenSIPS-Users] Decrease the priority of contact

t; users-requ...@lists.opensips.org >>> >>> You can reach the person managing the list at >>> users-ow...@lists.opensips.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Content

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.9.0 - a new major release is out

Can't wait to play with CouchBase and/or MongoDB. Sounds great! On Tue, Jan 29, 2013 at 2:20 PM, Saúl Ibarra Corretgé wrote: > > On Jan 29, 2013, at 8:11 PM, Bogdan-Andrei Iancu wrote: > > > Hi all, > > > > One more major release is out - OpenSIPS 1.9.0 Release Candidate (beta) > > > > *OpenSIP

Re: [OpenSIPS-Users] Decrease the priority of contact

I'm not 100% sure what you are trying to do. I needed to do some serial calling and here is what I did http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-Bugs-3349030-Need-access-to-SIP-URI-Q-Value-td6554597.html http://www.opensips.org/Resources/DocsCoreVar19#toc75 On Tue, Jan 29

Re: [OpenSIPS-Users] Passing Register requests with B2BUA Topology Hiding

You need to re-read the B2B_Logic module. http://www.opensips.org/html/docs/modules/1.8.x/b2b_logic.html#id292703 You should only call the b2b_init_request function on an initial invite. So you need something like this if(is_method("INVITE") && !has_totag() before you call b2b_init_request Yo

Re: [OpenSIPS-Users] Passing Register requests with B2BUA Topology Hiding

I don't see you doing any register functions in your script. On Friday, January 11, 2013, Jeryes . wrote: > Hi, > > I have a very simple b2bua server (running on opensips 1.7) that only uses > topology hiding to make the communication between two sip server, and it's > working fine to process cal

Re: [OpenSIPS-Users] Polycom REGISTER loop

s just the phones within the network that are > complaining. Looking at the NAT box, I see that they have 5060 port > forwarded to the OpenSIPS box. > Pinging the phone from the proxy is ok. > > Any thoughts? > > Nick. > > > > On 1/9/13, Duane Larson wrote: > >

Re: [OpenSIPS-Users] Polycom REGISTER loop

So you say you "closed gaping holes". Are you saying that the local polycom phone are behind a firewall? If they are behind the firewall then you need to figure out why the 200 OK is not making it to the polycom phones. It could be because Polycom doesn't support rport (just a shot in the dark f

Re: [OpenSIPS-Users] Get user_agent after lookup

eed to know user agent who will receive the local call. > The location table contain this information in user_agent field. > I need to receive this field after run lookup("location","m") . > > > 2012/12/20 Duane Larson : > > http://www.opensips.org/Resourc

Re: [OpenSIPS-Users] Get user_agent after lookup

http://www.opensips.org/Resources/DocsCoreVar18#toc92 Is that what you want? On Thu, Dec 20, 2012 at 2:59 PM, Dragomir Haralambiev wrote: > Hello All, > > I rum save("location") wnen reseive REGISTER. > Ho to get user_agent field from location table after run > lookup("location","m") ? > > if (

Re: [OpenSIPS-Users] modules/db_mysql help

OpenSIPS doesn't need to connect to MySQL depending on what you want to do with OpenSIPS and the modules you want to use. For example... Look at the auth_db module http://www.opensips.org/html/docs/modules/1.8.x/auth_db.html If you are going to have users and passwords then you would need this m

Re: [OpenSIPS-Users] CDRTool - Unknown module "Connect-Info"

#x27;, \ '%{Connect-Info}', \ '%{X-RTP-Stat}', \ '%{Acct-Session-Id}', \ '%{Sip-To-Tag}', \ '%{Sip-From-Tag}' \ )" On Fri, Nov 9, 2012 at 8:33 P

Re: [OpenSIPS-Users] CDRTool - opensips.radacct201211 doesn't exist

ary > and the mysql server could cause this. > > On Nov 8, 2012, at 10:34 PM, Duane Larson wrote: > > I'm at a loss. I upgraded everything on the server, php and all, and that > didn't really help. I no longer get the "MySQL error: 1146 (Table > 'opensips.ac

Re: [OpenSIPS-Users] OpenXCAP - Can't connect to MySQL server when mysql port is defined

base on a port other than 3306. Thanks for the quick response and helping me with the bug! On Thu, Nov 8, 2012 at 3:08 AM, Saúl Ibarra Corretgé wrote: > Hi Duane, > > On Nov 7, 2012, at 6:22 PM, Duane Larson wrote: > > > I just moved over to a new mysql server and the port I need to

Re: [OpenSIPS-Users] CDRTool - opensips.radacct201211 doesn't exist

bal.inc is set to look in the radius database?? On Thu, Nov 8, 2012 at 12:18 PM, Adrian Georgescu wrote: > I recall seeing this myself once. Then it was caused by a broken or > mismatched with the server version PHP mysql client library. Upgrading it > solve the issue at that time.

Re: [OpenSIPS-Users] CDRTool - opensips.radacct201211 doesn't exist

ing I changed the config to use 19994 and redirected the port > 19994 to 3306 on the mysql server. I got no errors an all was working. > > > Best regards, > > -- > Tijmen de Mes > AG Projects > > > On 11/07/2012 05:27 PM, Duane Larson wrote: > > Ok. Just to tes

[OpenSIPS-Users] OpenXCAP - Can't connect to MySQL server when mysql port is defined

I just moved over to a new mysql server and the port I need to connect to is no longer the default 3306 that mysql uses. I now need to connect to port 19994 on the new mysql server. With OpenXCAP are you able to connet to define a different mysql port? When I restart OpenXCAP I see the following

Re: [OpenSIPS-Users] Release 1.9 Feature Request: REST queries

Brett, I almost thing there needs to be a new thread to discuss NoSQL. I don't have any experience with cassandra, Redis, MongoDB or Couch but I keep hearing more and more about them. I believe Adrian Georgescu mentioned MongoDB and that appears to have "JSON-style documents". On Wed, Nov 7, 2

Re: [OpenSIPS-Users] CDRTool - opensips.radacct201211 doesn't exist

rver. On Wed, Nov 7, 2012 at 9:18 AM, Tijmen de Mes wrote: > Hi, > > Ok, I can't see quickly where it goes wrong, in my opinion it should just > work. > > Tomorrow I can do some tests to see if I can track where it goes wrong. > > > Best regards, > > -- > Tijmen de

Re: [OpenSIPS-Users] CDRTool - opensips.radacct201211 doesn't exist

ssword"; var $Halt_On_Error ="yes"; } On Wed, Nov 7, 2012 at 8:33 AM, Tijmen de Mes wrote: > Hi Duane, > > So to narrow it down, CDRTool is able to connect to the db ( you see them > in the processlist?), but then it looks in the wrong databases? > > Best regard

[OpenSIPS-Users] CDRTool - opensips.radacct201211 doesn't exist

I have been running CDRTool for a long time (about 2 or 3 years) and today I moved the mysql database over to a new server. For some reason when I log into CDRTool I see the following at the bottom of the screen MySQL error: 1146 (Table 'opensips.active_sessions' doesn't exist) Session halted. t

Re: [OpenSIPS-Users] CDRTool - Purge Mediaproxy's media_sessions records

Thanks Adrian. Was wondering why it was getting so big :) On Nov 6, 2012 2:53 AM, "Adrian Georgescu" wrote: > There is no automatic purge mechanism in place, you can easily script a > cron job to purge them using a mysql query. > > Adrian > > > On Nov 5, 2012,

[OpenSIPS-Users] CDRTool - Purge Mediaproxy's media_sessions records

I have a good amount of rows in my "media_sessions" table and was curious how these fields get purged. I see that with sip_trace you can do "purgeRecordsAfter" => "7". Is this possible with media_trace or does media_trace depend on the "purgeCDRsAfter" setting that you configure for your "opensi

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

cases (when call comes form GW with IP X > and when you force routing to GW with IP X). > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 11/05/2012 05:27 PM, Duane Larson wrote: > > Ali, > > Jus

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

Ali, Just to add to the scenarios when you don't use load balancer but want to keep up with load. If you are load balancing to Asterisk servers but you don't use the load_balance() function because you want an "attended transfer" or someone is call for a "Parked Call". You want to make sure

Re: [OpenSIPS-Users] [RELEASES] Planing OpenSIPS 1.9.0 major release

Good point. On Fri, Oct 26, 2012 at 10:54 AM, Saúl Ibarra Corretgé wrote: > > On Oct 26, 2012, at 5:46 PM, Duane Larson wrote: > > > Is there any roadmap for "SIP over Websocket"? I know there is now > OverSIP but wasn't sure if OpenSIPS had any plans to i

Re: [OpenSIPS-Users] [RELEASES] Planing OpenSIPS 1.9.0 major release

Is there any roadmap for "SIP over Websocket"? I know there is now OverSIP but wasn't sure if OpenSIPS had any plans to implement a module. Just asking since WebRTC is still evolving. On Fri, Oct 26, 2012 at 10:34 AM, Ovidiu Sas wrote: > Locks for config could be integrated in cfgutils (inst

Re: [OpenSIPS-Users] Broken Link in cook books.

http://opensips-open-sip-server.1449251.n2.nabble.com/Python-module-on-OpenSIPS-td5018382.html#a7581444 On Thu, Oct 11, 2012 at 1:57 AM, qasimak...@gmail.com wrote: > Hi, > > Just wondering why the following link is not available in documentation. > > http://www.opensips.org/html/docs/modules/dev

Re: [OpenSIPS-Users] B2BUA - Transfer Issue because of no SDP

On Mon, Oct 8, 2012 at 3:35 AM, Binan AL Halabi wrote: > Hi Duane, > > Try to add SDP body manually if it is not added: > - Save the original SDP ($rb) somewhere before calling b2b_init_request() > function. > - Add it to reInvite in local_route. > > //Binan > --

Re: [OpenSIPS-Users] call a standard phone and mobile at the same time

Can you explain in more detail what you are trying to do? On Mon, Oct 8, 2012 at 8:32 AM, Engineer voip wrote: > Hello, > I do that and its good but if user A hangs up the call, the user B is > hanging too. > I want when user A hangs up or responder, the user B can take the call! > have you an

Re: [OpenSIPS-Users] avpops module and opensips.cfg ?

Very good book http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book On Mon, Oct 8, 2012 at 8:12 AM, Pierre-Yves Marche wrote: > Hello opensips community, > > I'm quite new to the community > > I'm trying to use "avpops" module to store & retrieve profile > information from

[OpenSIPS-Users] B2BUA - Transfer Issue because of no SDP

I have a couple of issues with transferring calls when my SIP provider is involved. Issue #1 - Call comes from SIP Provider to Customer A100. Customer A100 picks up. Customer A100 transfers call to Customer A102. While Customer A102's phone is ringing the SIP Provider user on the line doesn't h

Re: [OpenSIPS-Users] Maximun calls for each user

Hi, > Yes, but what i do if i want to define a number of calls for each user? > For example: 5 calls of user A and 10 calls for user B. > > 2012/10/7 Duane Larson > >> I think this is what you want. Since you are using OpenSIPS 1.8 some >> things might be a little differe

Re: [OpenSIPS-Users] Maximun calls for each user

I think this is what you want. Since you are using OpenSIPS 1.8 some things might be a little different but not by much. http://www.opensips.org/Resources/DocsTutConcurrentCalls On Sun, Oct 7, 2012 at 1:45 PM, Engineer voip wrote: > Hello All, > I use opensips 1.8 and i want to limite a maxi

Re: [OpenSIPS-Users] bash shell variable not kept when run in opensips.cfg

It sounds like you are trying to reinvent the wheel. Why not just use the http://www.opensips.org/html/docs/modules/1.8.x/db_mysql.html module so you can query your database. And then like Ali says you can use AVPOPS module with a little scripting in the OpenSIPS config. You really don't need to

Re: [OpenSIPS-Users] b2b preserve header.

Read this http://www.opensips.org/html/docs/modules/1.8.x/b2b_logic.html#id250020 Does that help or is that not what you are looking for? On Wed, Sep 12, 2012 at 5:26 AM, Jorge Henrique Pinho < jorge-h-pi...@ext.ptinovacao.pt> wrote: > Hi all > In b2b, is there a way to preserve headers added in

Re: [OpenSIPS-Users] Python module on OpenSIPS ?

(as usage). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 08/25/2012 07:39 PM, Duane Larson wrote: The python module is no longer maintained and is now obsolete. Bogdan greyed out the link a while back. On Aug 25, 2012 9:01 AM

Re: [OpenSIPS-Users] Python module on OpenSIPS ?

The python module is no longer maintained and is now obsolete. Bogdan greyed out the link a while back. On Aug 25, 2012 9:01 AM, "shaahin" wrote: > Hi Maksym, I tried finding the documentation on the CookBooks > webpage, > but there still is no

Re: [OpenSIPS-Users] Get value from the Presence module table

Pieri, I am guessing there are more scenario's than "somebody is trying to call Bob and its status is 'busy', then send the call to another entity", but at least with the Bob scenario you could use the Dialog module and keep up with how many calls the CALLER has going and fail over to a dif

Re: [OpenSIPS-Users] TM - possible mi.c issue

the cseq (neither the increment value) 2) if you do not want opensips to use default, simply push your own cseq via the "headers" param. Best regards, Bogdan Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 08/16/2012 06:56 AM, Du

Re: [OpenSIPS-Users] httpd module not working

Thanks for the help. Setting pkg to (-M 8) worked. On , Ovidiu Sas wrote: The httpd module needs to reserve a buffer for building http responses. If there are a lot of modules loaded, the pkg memory might become fragmented and there will be not enough space to allocate a big buffer.

Re: [OpenSIPS-Users] httpd module not working

I ran the sample OpenSIPS config that comes with all installs and I was able to start OpenSIPS with the httpd and mi_http modules. Not sure why its not starting with my config. I will send you the debug to you directly after I send this email. On , Ovidiu Sas wrote: Can you enable debug p

[OpenSIPS-Users] httpd module not working

I wanted to try out the mi-http module but I am not able to get OpenSIPS to start up when I have enabled the httpd module. When I start OpenSIPS I am seeing the following error Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]: NOTICE:presence:child_init: init_child [-2] pid [1537]

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

ameters. > > > > So, from what I see, that TO header is correct. > > > > > > > > Regards, > > Vlad Paiu > > OpenSIPS Developer > > http://www.opensips-solutions.com > > > > On 08/09/2012 06:13 AM, Duane Larson wrote: > > > > > &

[OpenSIPS-Users] Incomplete REFER sent using mi_xmlrpc with t_uac_dlg

I am trying to generate a REFER message using mi_xmlrpc and the t_uac_dlg MI Function. When I send the xmlrpc info to the OpenSIPS server on port it receives the info but for some reason it sends the REFER message without the "Contact" or "Refer-To" headers included. I am not sure why or how

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

considered to be SIP URI parameters. So, from what I see, that TO header is correct. Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 08/09/2012 06:13 AM, Duane Larson wrote: I changed the following in the ctd.sh script Changed the default of &

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

n : are you sure your opensips is > > listening on the given IP:port ? have you check with netstat ? > > also have you checked with netstat also if there is traffic queued > > on the sockets ? > > > > > > > > Regards, > > > > > > Bogdan-Andre

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

traffic queued on the sockets ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/27/2012 12:48 AM, Duane Larson wrote: Oh yeah. My first email has the SIPTrace from the OpenSIPS/SBC. So I am logged into the OpenSIPS/SBC and did the NGREP. So

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

Oh yeah. My first email has the SIPTrace from the OpenSIPS/SBC. So I am logged into the OpenSIPS/SBC and did the NGREP. So I see it SIP invite (99.XX.XX.161 is the IP of the OpenSIPS/SBC). I would even let you log into the OpenSIPS/SBC and see it for yourself. Makes no sense. U 2012/07/19 18

Re: [OpenSIPS-Users] B2B from heading missing caller name

The solution from what I remember was a bug and it was fixed in the trunk. I can't be 100% with the timeline but the 1.8 release might have it in there. 1.8 was release in March 2012 and the bug was found in March 2012 so no telling. Ovidiu might remember the the bug tracker on sourceforge or mig

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

Hey Bogdan, I think you might be a little confused from my emails. The last email that had a SIP trace and the 100 Trying was a click-to-dial generated from php-sip(http://code.google.com/p/php-sip/) and I wanted to show you that with php-sip the OpenSIPS server processes the INVITE and rep

Re: [OpenSIPS-Users] b2b and Cancel message!

In what route are you using "if(is_method("CANCEL"))"? You should be looking in the route[b2b_reply] {} I think. http://www.opensips.org/Resources/B2buaTutorial#toc19 On , Jorge Henrique Pinho wrote: Hello, i am using b2bua module with topology hiding, and I cannot filter the 200OK and A

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

That is correct there is no answer to the INVITE and it doesn't appear to even enter the main route of the OpenSIPS config. I have xlogs set up in the opensips script and I never see the INVITE enter. Here is another sip trace from the php-sip click to call program and for some reason this

[OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

Has anyone used the ctd.sh example that comes with Opensips in the "example" folder? I am trying to use it and the INVITE gets sent out but nothing happens. I even tried with sending the INVITE to an OpenSIPS server and the OpenSIPS server never even sees it enter the main route even though

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

Just to update... My config on the OpenSIPS/SBC was all jacked up. I basically have two routes in my config, one for SIP messages coming from the LAN and one for SIP messages coming from the WAN. On the LAN side before I was doing my "if has_totag" and "if loose_route" I had unintentionally pu

Re: [OpenSIPS-Users] problem with loose_route()/t_relay() across multiple interfaces

It hasn't been resolved but I was able to everything work. It looks like it has to do with the INVITE that is being delivered to the Callee. The Record-Routes are in the wrong order. I was able to remove all Record-Routes from the INVITE, reorder them in the correct order and then send the

Re: [OpenSIPS-Users] problem with loose_route()/t_relay() across multiple interfaces

Jeff, Read my post. I got my calls working but I am not sure if it was a bug or something is just jacked up with my config and I am not following the RFCs the right way. I can go into more detail offline if you need. I hope to get some feedback on my post. On , Jeff Pyle wrote: Hi Duane

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

I just got my calls working by removing the Record-Route's and then reinserting then in an order that would according to my topology. I will need to go back and start from scratch to see if a lot of the other stuff I did was really needed or not and then update but here is were I edited the

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

I think I have multiple issues going on but I might be getting closer to the issue. I am wondering if this might be part of the issue. If you look at the the following, http://www.tech-invite.com/Ti-sip-dialog.html#inv , for the first INVITE message that the Callee receives the first Proxy

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

I'm really not sure if I am just duck taping the issue but I was able to make most of the call work. The only problem now is when the Callee hangs up the BYE is sent directly to the OpenSIPS/Proxy IP instead of going to the OpenSIPS/SBC. This will not work due to firewall issues. My ACKs ar

Re: [OpenSIPS-Users] problem with loose_route()/t_relay() across multiple interfaces

Almost sounds like you and I are having the same issue. Here's my issue http://opensips-open-sip-server.1449251.n2.nabble.com/Two-OpenSIPS-proxies-issue-td7580685.html Do you have a SIP trace? I'm just wondering if we are having the same problem. Does the ACK that gets relayed to ifself on the

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

ne.lar...@gmail.com> wrote: > > > > > > I see that the third 200 OK is edited by my OpenSIPS/Proxy server but > that is because in the location table in the "received" field I stored the > received Public IP address so that the replies are sent to the Public IP > address of the device instead of the private IP

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

P address so that the replies are sent to the Public IP address of the device instead of the private IP that was in the original contact header. > > > > > > > > On , Ali Pey ali...@gmail.com> wrote: > > > Examine the Contact header of the 200 OK. That's where this usu

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

Ali Pey ali...@gmail.com> wrote: > Examine the Contact header of the 200 OK. That's where this usually gets messed up. > > Regards, > Ali Pey > > On Wed, Jul 4, 2012 at 3:49 PM, Duane Larson duane.lar...@gmail.com> wrote: > > >

Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

e original contact header. On , Ali Pey wrote: Examine the Contact header of the 200 OK. That's where this usually gets messed up. Regards, Ali Pey On Wed, Jul 4, 2012 at 3:49 PM, Duane Larson duane.lar...@gmail.com> wrote: I have the following scenario LAN OpenSIPS/SBC

[OpenSIPS-Users] Two OpenSIPS proxies issue

I have the following scenario LAN <-> OpenSIPS/SBC <-> INTERNET <-> OpenSIPS/Proxy I have the OpenSIPS/SBC device because the firewall that is protecting the LAN doesn't play well with SIP. It has the following IPs (LAN = 192.168.88.1), WAN (99.xx.xx.161). The OpenSIPS/Proxy device sits on th

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