Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

excuse me, posted message to wrong thread -- best regards, Alexey https://alexeyka.zantsev.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

Is it possible to cache not exact number of table row but all values from concrete column from all rows? The total amount of rows in table is not big, several rows or several dozens. -- best regards, Alexey https://alexeyka.zantsev.com/ ___ Users mail

Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

I want to setup a call center. Sent from Outlook for Android<https://aka.ms/AAb9ysg> From: Users on behalf of Alexey Sent: Friday, February 9, 2024 12:24:11 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integ

Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

Hello, What exact kind of integration do you mean? Any peculiarities of the 18.20 release? There is a tutorial [1] but for OpenSIPS 1.8 and Asterisk 1.8. So, the main difference according to Asterisk is moving from chan_sip towards chan_pjsip. [1] https://www.opensips.org/Documentation/Tutorial

[OpenSIPS-Users] OpenSIPS - Asterisk Integration

Has anyone done the integration of Opensips 3.4 with Asterisk 18.20? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] OpenSIPs -> Asterisk

Hello, I'm looking for the best configuration for this setting: OpenSIPs -> AsteriskPBX on a local server without public IP and with remote and local extensions. Testing different configurations I can call from remote extension and access to media services configured on Asterisk. I can mak

[OpenSIPS-Users] OpenSIPS - Asterisk issue

Hi, I am integrating OpenSIPS and Asterisk to use Asterisk to play media (typical media treatment) I have a softphone registered to OpenSIPS and when i call a specific number, a simple prompt needs to be played from asterisk. I have the sip configuration and also extensions.conf file setup. When

Re: [OpenSIPS-Users] OpenSIPS-Asterisk Integration

Hi all, To integrate OpenSIPS with Asterisk, both should be in same server? or If we can install OpenSIPS ans Asterisk in two different servers, how can I connect those two? Could any one please tell me, where is the configuration for that? Thanks in advance On Wed, Oct 9, 2013 at 10:57 AM, Si

[OpenSIPS-Users] OpenSIPS-Asterisk Integration

Hi, I am new to OpenSIPS, and trying to integrate OpenSIPS with Asterisk. I am successfully installed OpenSips in my machine using this tutorial. I have my Asterisk & OpenSIPS in two different servers. How Can I Connect these two. Where do I nee

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Nick ... i`m making some tests, and I changed this block in my opensips.cfg and WORKED with any domain I define. # authenticate the REGISTER requests if (!www_authorize("10.1.1.2", "subscriber")) { www_challenge("10.1.1.2", "

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Hi Nick ... I have tried with your modparam set but no look. If I register with dns name, the register not work. What I have? maybe help you canhelp me with what I need ;) I have Asterisk to handle registration, routes and billing, but my customer base are increasing, and I want use Opensips to

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

There is also: 1. modparam("auth_db|usrloc|uri", "use_domain", 1) Please change that to 0. It's been a while since I have dealt with REGISTER authentiacation issues. Are you sure you need it? It's quite a resourceful process as the number of clients increase. What we do now, is use: 1) The

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

I think if I can change this query, it will work: Jul 18 18:00:13 opensips /usr/sbin/opensips[7146]: DBG:db_mysql:re_init_statement: query is , ptr=(nil) Use without domain= and now accept any domain. It is possible? Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 20

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Sorry ... for test.provider.com I use opensips.provider.com.br ;) Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/18 Willian Mazzardo - SYSSVOIP > Hi Nick ... thanks for your help ... very appreciated. > > Here is the news... changed the opensips.cfg to your mod

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Hi Nick ... thanks for your help ... very appreciated. Here is the news... changed the opensips.cfg to your modeparams, and use_domain to 0 and the problem still the same: In my domain tables I have all domains (10.1.1.2, test.provider.com) but into subscriber, i have recreated without VIEW and u

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

First things first. Forget the asterisk box for now. The phones are not getting registered with OpenSIPS. dOES the username, domain, in the digest make sense? Also, recheck the auth part of your configuration: # - auth params - modparam("auth","username_spec","$avp(user_spec)") modparam("a

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

This is my sip trace: http://pastebin.com/jALwj12r Scenario: 10.0.0.3 = Sip soft phone (behind NAT) 10.1.1.2 = Opensips My subscriber table: [image: Imagem inline 1] If in subscriber table I have in DOMAIN = 10.1.1.2 it works nice. And there is my question. Not all my customers use same dn

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

My billing system require asterisk 1.4. Did you check my opensips.cfg to see if everything is ok about handle register messages? Thanks Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Nick Khamis > Is it possible to use 1.8? I am not familiar with Asterisk 10

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Very strange situation I have with this scenario ... Most of registry coming into opensips are rejected ... returning Proxy Authentication Require .. I have some domains registered into mysql table... like dns name ( sip.provider.com) and External IP ( 222.222.222.222) and my internal ip (10.

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Here is my configs: opensips.cfg : http://pastebin.com/eLh2fUet My scenario: Opensips = 10.1.1.2 Mysql = 10.1.1.249 Asterisk = 10.1.1.247 Im trying to use balancing mode, but when I try to register into Opensips, return Method Not Allowed... here is the opensips.cfg for balancing: http://pa

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Hi ... im trying again ... and now WORKED !! ;) Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Willian Mazzardo - SYSSVOIP > No ... just sip messages, and stops at Proxy Authentication Require. > > Willian Mazzardo > Depto TI - SYSSVOIP > www.syssvoip.com.br >

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

No ... just sip messages, and stops at Proxy Authentication Require. Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Dani Popa > when you send a call in asterisk, do you see in asterisj cli that call hit > you callingcard context or it hit default context ? > >

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

when you send a call in asterisk, do you see in asterisj cli that call hit you callingcard context or it hit default context ? On Wed, Jul 17, 2013 at 1:55 PM, Willian Mazzardo - SYSSVOIP < will...@syssvoip.com.br> wrote: > My a2billing context > > [callingcard] > > exten => _X.,1,DeadAGI(a2bill

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

My a2billing context [callingcard] exten => _X.,1,DeadAGI(a2billing.php) Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Dani Popa > what contex hit invite from opensips ? > > > On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazzardo - SYSSVOIP < > will...@syssvoip

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

what contex hit invite from opensips ? On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazzardo - SYSSVOIP < will...@syssvoip.com.br> wrote: > Hi Dani ... thanks ... i have for now insecure=very ... my asterisk > version is 1.4... and this type of setting is for 1.6+ > > Willian Mazzardo > Depto TI - S

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Hi Dani ... thanks ... i have for now insecure=very ... my asterisk version is 1.4... and this type of setting is for 1.6+ Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Dani Popa > set opensips peer to insecure=port,invite > > > On Wed, Jul 17, 2013 at 1:12 PM

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

set opensips peer to insecure=port,invite On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP < will...@syssvoip.com.br> wrote: > Hi Stephens... how do I do this? > > Willian Mazzardo > Depto TI - SYSSVOIP > www.syssvoip.com.br > 55 3537 2030 > > > 2013/7/17 Stephen Vigus > >> Hi Willi

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Hi Stephens... how do I do this? Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Stephen Vigus > Hi Willian > > You most likely need to configure Asterisk to not authenticate SIP > requests coming from Opensips. > > Regards > Stephen > > > > On Wed, Jul 17, 2013

Re: [OpenSIPS-Users] Opensips + asterisk 1.4

Hi Willian You most likely need to configure Asterisk to not authenticate SIP requests coming from Opensips. Regards Stephen On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP < will...@syssvoip.com.br> wrote: > Hi all.. > > I know this is a very simple scenario, all PSTN calls be r

[OpenSIPS-Users] Opensips + asterisk 1.4

Hi all.. I know this is a very simple scenario, all PSTN calls be routed to asterisk to do the billing job, but im having some problems, this is my scenario: Sip Client (10.0.0.3) > Opensips (10.1.1.2) > Asterisk (10.1.1.247) . > PSTN Calls between sip clients on Opensips are working, but wh

Re: [OpenSIPS-Users] Opensips + Asterisk on the same server

To All, who read this. Flavio was right about similar IP, I found that .231 IP, that I used for test as separate IP was in the "domain" table and opensips recognized it as local and made loop. So for all - if you are doing schema where opensips and asterisk are on the same host - use different IP

Re: [OpenSIPS-Users] Opensips + Asterisk on the same server

I checked this on different IP addresses and there is a problem: when I use record_route and and loos_route, subsequent ACKs and BYEs are being looped. But when I do this: if ($DLG_status!=NULL) { if (!validate_dialog()) { fix_route_dialog(); } } after loos_ro

Re: [OpenSIPS-Users] Opensips + Asterisk on the same server

Flavio, thanks for the reply! I accidentally disabled email delivery, just noticed your response Well, it is bad. Bogdan, probably you know a good solution to avoid this problem?? I found lots of similar problems in google, but all of them are unresolved how they suppose to be. I see one solu

Re: [OpenSIPS-Users] Opensips + Asterisk on the same server

Hi Roman, I had the same problem and in all the cases I've used a new IP address for Asterisk. The problem is related to the routing of sequential messages, if the address is the same (not considering the port) it considers the destination as itself and loops the sequential requests. I'm not sure

[OpenSIPS-Users] Opensips + Asterisk on the same server

Hello! I am having problems with relaying messages. And I guess it is related to the ports in From/To headers. My schema is : UAC <-> ( OpenSips:5060 <-> Asterisk:5061 ) Opensips and asterisk are on the same host (it is a schema requirement - I have another similar mirror). Asterisk works as

Re: [OpenSIPS-Users] opensips + asterisk,

2013 11:48:43 To: OpenSIPS users mailling list Reply-To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] opensips + asterisk, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] opensips + asterisk,

Dear all, i have integrated asterisk with opensips server with the help of these link, http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration my voip clients are registered well,and there is audio on both the sides in WiFi network. but the same configuration is not working under

Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

n. > > > > Regards, > > > > Aamir Chougule > > Cell: 09167989111 > > > > From: Olle E. Johansson > > To: aamir chougule ; OpenSIPS users mailling list < > users@lists.opensips.org> > > Sent: Monday, 2 July 2012 7:08 PM > > Subje

Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

; > Regards, > > Aamir Chougule > Cell: 09167989111 > > From: Olle E. Johansson > To: aamir chougule ; OpenSIPS users mailling list > > Sent: Monday, 2 July 2012 7:08 PM > Subject: Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new wa

Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

: 09167989111 From: Olle E. Johansson To: aamir chougule ; OpenSIPS users mailling list Sent: Monday, 2 July 2012 7:08 PM Subject: Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way 2 jul 2012 kl. 13:34 skrev aamir chougule: > Wanted S

Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

2 jul 2012 kl. 13:34 skrev aamir chougule: > Wanted Scenario: > > Calls comes in to OpenSIPS server ==> Authentication & Proxying part will be > done by OpenSIPS ==> Call is relayed to Asterisk Server ==> Asterisk Server > provides the IVR services to fetch the number from the customer ==> Ast

Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

-Users] OpenSIPS & Asterisk Integration in a new way First of this is not new.  Just in your asterisk servers where you define the carriers replace all the carriers with just one opensips peer. Then in opensips.cfg on any incoming call detect that the call is coming from your asterisk server&#x

Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

First of this is not new. Just in your asterisk servers where you define the carriers replace all the carriers with just one opensips peer. Then in opensips.cfg on any incoming call detect that the call is coming from your asterisk server's - if that turns out to be yes use the LCR module to send o

[OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

Hi Team, I have installed opensips and asterisk in this way as given below: Current Scenario: Calls comes in to OpenSIPS server ==> Authentication & Proxying part is done by OpenSIPS ==> Call is relayed to Asterisk Server ==> Asterisk Server ask the customer to dial the number that he/she wan

Re: [OpenSIPS-Users] opensips - asterisk registrations

Hi, I think this step is also mentioned in any tutorial for integrating asterisk with opensips. But anyway don't forget to reply the exact config set which worked for you here for others to get help from. Regards, Sammy On Mon, Dec 5, 2011 at 8:49 PM, Matt Hamilton wrote: > > >You can add a Open

Re: [OpenSIPS-Users] opensips - asterisk registrations

>You can add a OpenSIPS peer in you asterisk sip.conf without any username and >password with param "insecure=invite,port" >that way your asterisk would not worry about asking for credentials on >incoming INVITES from opensips. That's what I was looking for, thanks for both of your answers.

Re: [OpenSIPS-Users] opensips - asterisk registrations

Hi, You can add a OpenSIPS peer in you asterisk sip.conf without any username and password with param "insecure=invite,port" that way your asterisk would not worry about asking for credentials on incoming INVITES from opensips. Regards, Sammy. On Sun, Dec 4, 2011 at 9:28 PM, Matt Hamilton wrote:

[OpenSIPS-Users] opensips - asterisk registrations

I'm a little confused about Opensips doing the registrations for Asterisk. I integrated Opensips with Asterisk Realtime (Asterisk sipusers/peers point to Opensips subscribe table via a view). Opensips does the authorization and saves the location. However, when a call comes in (INVITE is route

Re: [OpenSIPS-Users] openSips - Asterisk and Session Timers: ACK is sent to 192.168.1.10

Hi, haven't check your trace, but a fast guess is that you do not fix the contact of the 200 OK re-INVITE is no fixed and carries back to asterisk a private contact. So, do fix_nated_contact() for the replies coming from behind a NAT too.. Regards, Bogdan wrote: > Hi, > > My setup: > - 11

[OpenSIPS-Users] openSips - Asterisk and Session Timers: ACK is sent to 192.168.1.10

Hi, My setup: - 11.22.33.44 : openSIPS 1.6.3 - 11.22.33.45 : one of the Asterisk 1.6.2.13 servers - 88.77.66.55 : my public ip-address - 192.168.1.10 : my local ip-address (NAT) All is working well except Session Timers where the Re-Invite originates from Asterisk. I have a SIP trace ( http://p

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Andrew Pogrebennyk : > On 06.10.2010 17:25, Stefano Sasso wrote: >> So I can resolve dnatting i.e. port 5061 to .131 and 5062 to .132 and >> having in load_balancer >> 77.238.xx.yy:5061 and 77.238.xx.yy:5062? >> Am I right? > > Yes, this should help. It seems that asterisk will append bin

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

On 06.10.2010 17:25, Stefano Sasso wrote: > So I can resolve dnatting i.e. port 5061 to .131 and 5062 to .132 and > having in load_balancer > 77.238.xx.yy:5061 and 77.238.xx.yy:5062? > Am I right? Yes, this should help. It seems that asterisk will append bindport to externip automatically now so

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Andrew Pogrebennyk : > This ACK should reach the asterisk: > U 2010/10/06 14:43:42.736777 192.168.6.130:5060 -> 77.238.yy.zz:5060 > ACK sip:77.238.yy.zz:5060;lr;ftag=931ba062;did=12c.0478d917 SIP/2.0. > ... > but then there is another ACK to itself. > > Are you doing NAT 77.238.yy.zz to 1

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

On 06.10.2010 16:36, Stefano Sasso wrote: > nothing happened. > It still loops (ACKs and BYEs) Hm, I will have to check in detail what you wrote here. This ACK should reach the asterisk: U 2010/10/06 14:43:42.736777 192.168.6.130:5060 -> 77.238.yy.zz:5060 ACK sip:77.238.yy.zz:5060;lr;ftag=931ba06

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Andrew Pogrebennyk : > Stefan, > Please try removing ip addr and domain of opensips from domains table. > It is sufficient to have listen=ip and alias="domain" lines in config. > Domain module will learn the ip and domain from config automatically. nothing happened. It still loops (ACKs

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

Stefan, Please try removing ip addr and domain of opensips from domains table. It is sufficient to have listen=ip and alias="domain" lines in config. Domain module will learn the ip and domain from config automatically. On 06.10.2010 16:02, Stefano Sasso wrote: > Hi Andrew, > thank you for the r

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Andrew Pogrebennyk : > Right, Stefano: make sure you have not added the opensips IP addresses > or domain names already listed in "alias" core parameter to the domain > table. If the address in RURI is considered local it does routing "after > strict". The RURI gets rewritten with the URI

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

On 06.10.2010 15:30, Vallimamod ABDULLAH wrote: > You are right: you should not mix record_route_preset() and record_route(). > Try to replace record_route with record_route preset. And if it does not > work, make a ngrep capture on your opensips server to see sip dialog between > opensips and as

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Vallimamod ABDULLAH : > You are right: you should not mix record_route_preset() and record_route(). ok > Try to replace record_route with record_route preset. And if it does not > work, make a ngrep capture on your opensips server to see sip dialog between > opensips and asterisk (com

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

On Oct 6, 2010, at 2:06 PM, Stefano Sasso wrote: > 2010/10/6 Vallimamod ABDULLAH : >>> Can this be resolved using advertised_address in opensips? or there is >>> other options? >> >> Then you should use record_route_preset with the public ip at the beginning >> of your script. > > thank you for

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Stefano Sasso : > 2010/10/6 Vallimamod ABDULLAH : >>> Can this be resolved using advertised_address in opensips? or there is >>> other options? >> >> Then you should use record_route_preset with the public ip at the beginning >> of your script. > > thank you for the hint. > I now have re

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Vallimamod ABDULLAH : >> Can this be resolved using advertised_address in opensips? or there is >> other options? > > Then you should use record_route_preset with the public ip at the beginning > of your script. thank you for the hint. I now have record_route_preset, and now the ACK fro

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

On Oct 6, 2010, at 1:25 PM, Stefano Sasso wrote: > 2010/10/6 Vallimamod ABDULLAH : >> It is asterisk that is not receiving the ACK so the issue is on your >> opensips config. >> Can you make a ngrep trace of an invite to see where is sent the final ACK >> from opensips ? More precisely, check if

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Vallimamod ABDULLAH : > It is asterisk that is not receiving the ACK so the issue is on your opensips > config. > Can you make a ngrep trace of an invite to see where is sent the final ACK > from opensips ? More precisely, check if the UAC sends the ACK to Opensips' > public IP and not

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

Actually, After reading back the logs: > [Oct 6 10:29:54] WARNING[25602]: chan_sip.c:3805 retrans_pkt: Hanging up > call NjZjMmI2MWRlYmY0YWYwMGVhYTAyNmE0NzU4OWU5YTk. - no reply to our > critical packet (see doc/sip-retransmit.txt). It is asterisk that is not receiving the ACK so the issue is on

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

2010/10/6 Vallimamod ABDULLAH : > Hi Stefano, Hi, > Make a sip trace on your asterisk box to see where the ACK is sent. Maybe you > need to enable nat on asterisk to force it to send the ACK to the originating > IP and not the IP of the contact field. Have a look at > http://www.voip-info.org/

Re: [OpenSIPS-Users] opensips+asterisk: signalling not working?

Hi Stefano, The 20s timeout is typically an ACK timeout (as someone reminded it on the list some time ago.) The asterisk log you pasted confirm it: the final ACK never reaches back Opensips so the dialog is cut down after the timeout. Make a sip trace on your asterisk box to see where the ACK i

[OpenSIPS-Users] opensips+asterisk: signalling not working?

Hello folks, here I am again :) I have a setup where I use opensips as registration+proxy and asterisk as media gateway. When I place a call I can correctly hear the call audio (so RTP flow is ok), and the callee can hear my voice too, but after 20 seconds the call hangs up. In the asterisk logs

Re: [OpenSIPS-Users] OpenSIPS+asterisk: cannot place call

2010/9/27 Stefano Sasso : >> if (is_method("INVITE")) { >>        if (!ds_is_in_list("$si", "$sp")) ) /* if it's not from asterisk */ >>                if (!ds_select_dst("1", "5")) { >>             >> >>        } else >>                route(1); /* send it out */ > > I inserted what you said,

Re: [OpenSIPS-Users] OpenSIPS+asterisk: cannot place call

2010/9/27 Anca Vamanu : > Hi Stefano, Hi, > I suppose that you have the nat traversal handled also on opensips, > right? ( since the host part of the clients registration in asterisk > points to opensips). Then in this case it is normal that the flow of the how should I handle this? My only rout

Re: [OpenSIPS-Users] OpenSIPS+asterisk: cannot place call

Hi Stefano, I suppose that you have the nat traversal handled also on opensips, right? ( since the host part of the clients registration in asterisk points to opensips). Then in this case it is normal that the flow of the invite is the following: sip client -> opensips -> asterisk -> opensips (

[OpenSIPS-Users] OpenSIPS+asterisk: cannot place call

Hello, I started testing OpenSIPS with my asterisk cluster. For this testing configuration I'm using only one asterisk server of the pool. (the registration is done on asterisk server) The OpenSIPS configuration routing part is: route{ if (!mf_process_maxfwd_header("10")) {

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

t;> Thanks > > > > >>> Doug > > > > >>> > > > > >>> > > > > >>> On 2010/06/07

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

doubt/not clear that after IVR (meaning > when the > > > user is authenticated > > > >>> through IVR) who will handle all the > > transactions and > > > dialog (

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

t; > > >>> > > > >>> I have followed up this link and tried extending > the > > > opensips.cfg file to > > > >>> route call to Asterisk. > > > >>> > >

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

> >>> I want Opensips to handle all the transactions. > > >>> > > >>> Any insight on this ? > > >>> > > >>> Thanks, > > >>>

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

.opensips.org > <mailto:users-boun...@lists.opensips.org> [mailto:users- > <mailto:users-> > > boun...@lists.opensips.org <mailto:boun...@lists.opensips.org>] > On Behalf Of Premalatha Kuppan > > Sent: Monday, 07. June 2010 11:35 > &

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

>>> <mailto:sebastian.schum...@t-com.sk>> wrote: > >>>> > >>>> Hi Prem > >>>> > >>>> There is a good tutorial at > >>>> http://www.opensips.o

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

ransactions and > > dialog (OPENSIPS or > > >>> Asterisk ) ? > > >>> > > >>> I want Opensips to handle all the transactions. > > >>> > > >>> Any insight on this ? >

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

;>> >>> >>> Thanks Sebastian. >>> >>> >>> >>> I have followed up this link and tried extending the opensips.cfg >>> file to >>> >>> route call to Asterisk. >>> >>> >>> >>> I

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

fter IVR (meaning when the user is >> authenticated >> >>> through IVR) who will handle all the transactions and dialog (OPENSIPS >> or >> >>> Asterisk ) ? >> >>> >> >>> I want Opensips to handle all the transactions. &

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

I want Opensips to handle all the transactions. > >>> > >>> Any insight on this ? > >>> > >>> Thanks, > >>> Prem > >>> > >>> On Mon, Jun 7, 2010 at 3:15 PM, Schumann Sebastian > >>> wrote: > >>>> > >

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

On Mon, Jun 7, 2010 at 3:15 PM, Schumann Sebastian >>> wrote: >>>> >>>> Hi Prem >>>> >>>> There is a good tutorial at >>>> http://www.opensips.org/Resources/DocsTutAsterisk It does exactly what you >>>> need I assume. >>>>

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

t;>> http://www.opensips.org/Resources/DocsTutAsterisk It does exactly what >>> you need I assume. >>> >>> For details in writing and extending basic configuration, you can find >>> also the linked documentation there. >>> >>> Best re

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

Original Message- >> > From: users-boun...@lists.opensips.org [mailto:users- >> > boun...@lists.opensips.org] On Behalf Of Premalatha Kuppan >> > Sent: Monday, 07. June 2010 11:35 >> > To: OpenSIPS users mailling list >> > Subject: [OpenSIPS-Us

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

Monday, 07. June 2010 11:35 > To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration > > Hi, > > Can anyone guide me in building the Opensips and Asterisk Integration. > > I want to Use OpenSIPS as SIP PROXY (

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

@lists.opensips.org] On Behalf Of Premalatha Kuppan > > Sent: Monday, 07. June 2010 11:35 > > To: OpenSIPS users mailling list > > Subject: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration > > > > Hi, > > > > Can anyone guide me in building the Opensips a

Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

age- > From: users-boun...@lists.opensips.org [mailto:users- > boun...@lists.opensips.org] On Behalf Of Premalatha Kuppan > Sent: Monday, 07. June 2010 11:35 > To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration > > Hi, > > Can anyone gui

[OpenSIPS-Users] OPENSIPS+ASTERISK Integration

Hi, Can anyone guide me in building the Opensips and Asterisk Integration. I want to Use OpenSIPS as SIP PROXY (i.e all transactions and dialogs should be handled by Opensips) and Asterisk to do only IVR functionality. I appreciate if anyone can guide me in writing a routing logic from Opensips

Re: [OpenSIPS-Users] opensips + asterisk...

Hi Douglas, you may take a look at the Tutorial for realtime integration between opensips and asterisk: http://www.opensips.org/Resources/DocsTutAsterisk Regards, Bogdan Douglas Adami wrote: > hello guys! > > I'm new here. > > Recently with the increased use of VoIP communications, I decide

Re: [OpenSIPS-Users] opensips + asterisk...

On Wed, Nov 11, 2009 at 7:41 PM, Douglas Adami < douglas.ad...@rumodigital.com> wrote: > hello guys! > > I'm new here. > > Recently with the increased use of VoIP communications, I decided deepen it > and possibly integrate new services to my business. I have a private server > (VPS) for our hosti

[OpenSIPS-Users] opensips + asterisk...

hello guys! I'm new here. Recently with the increased use of VoIP communications, I decided deepen it and possibly integrate new services to my business. I have a private server (VPS) for our hosting / email and some domains, it was run Asterisk as a gateway to reduce the cost of the calls an

Re: [OpenSIPS-Users] opensips+asterisk call dropping in 20 seconds

On Mon, Jul 13, 2009 at 6:30 PM, Bogdan-Andrei Iancu wrote: > Hi Ram, > > a call drop (at signalling level) after 20 secs typically shows a problem > with the ACK - the ACK does not get back to callee and callee bye's the call > as it never think the call was not confirm. > > Could you post a tr

Re: [OpenSIPS-Users] opensips+asterisk call dropping in 20 seconds

Hi Ram, a call drop (at signalling level) after 20 secs typically shows a problem with the ACK - the ACK does not get back to callee and callee bye's the call as it never think the call was not confirm. Could you post a trace for the KO call ? Regards, Bogdan ram wrote: > Hi > > In continu