[OpenSIPS-Users] R: R: [OpenSIPS-Devel] next_branch function, NAT and PATH (RFC 3327)

Sorry, there is a mistake... If all contacts have the same q-value -Messaggio originale- Da: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] Per conto di Mauro Davi' Inviato: lunedì 18 maggio 2009 10:25 A: Bogdan-Andrei Iancu Cc: users@lists.opensip

[OpenSIPS-Users] R: [OpenSIPS-Devel] next_branch function, NAT and PATH (RFC 3327)

} else { xlog("L_INFO", "Service Unavailable - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); $avp(s:status_code)="503"; acc_db_reque

[OpenSIPS-Users] next_branch function, NAT and PATH (RFC 3327)

Hi All, I'm using a SIP Server connected to a SIP Proxy (load balancer) When a client behind NAT is connected to the SIP flatform all works fine, except one issue: If I call an AOR via parallel fork all goes fine... But if I use a serial forking, when I call the next_branch function to handl

[OpenSIPS-Users] R: Media Proxy between LAN and WAN

MD Da: Brett Nemeroff [mailto:br...@nemeroff.com] Inviato: mercoledì 13 maggio 2009 15:44 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: [OpenSIPS-Users] Media Proxy between LAN and WAN Mauro, This effect: c=IN IP4 0.0.0.0172.30.0.2

[OpenSIPS-Users] Media Proxy between LAN and WAN

Hi All, I have a Media Proxy installed in our VoIP platform. The Media Proxy machine have 2 Network card, one on a Private LAN the other on a WAN. Now I configured the Media-Relay with the IP address 0.0.0.0 to listen to all the interfaces. On opensips when then INVITE came from a UA direc

[OpenSIPS-Users] NAT e REGISTRAR Module

Hi All, I would that the opensips Registrar will process the REGISTER method in 2 different way: 1) The registration process use the default parameter (max expire time 3600 seconds) 2) If the client is behind a NAT the max expire interval will be set to 30 seconds, so the clien

[OpenSIPS-Users] Location table and received field

Hi All, I would to know how the save function store the received field in the location table. With same client this field is present correctly with the received SIP:IP:PORT value, but in some circumstances this field is NULL. Could anyone can tell me why? I have a SIP server behind

[OpenSIPS-Users] DRouting Module and LCR Module

Hi All, I need to route a call based on domain name. For example I need to route a call to a SBC if the domain is equal to 'domain.es'. The DRouting module and the LCR module can route a call based on username's rules. There is a way to route a call on a rule based on the domain part of

[OpenSIPS-Users] R: Ring Group - Memory Hunt - with serial forking?

OK.. now I understand!!! :-) -Messaggio originale- Da: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] Per conto di Chris Maciejewski Inviato: martedì 21 aprile 2009 14:06 A: users@lists.opensips.org Oggetto: Re: [OpenSIPS-Users] Ring Group - Memory Hunt - with

[OpenSIPS-Users] R: R: Ring Group - Memory Hunt - with serial forking?

on...)? Regards, MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: martedì 21 aprile 2009 11:11 A: Mauro Davi' Cc: Chris Maciejewski; users@lists.opensips.org Oggetto: Re: R: [OpenSIPS-Users] Ring Group - Memory Hunt - with serial fo

[OpenSIPS-Users] R: Ring Group - Memory Hunt - with serial forking?

Sorry Bogdan, If I understand, this is the scenarious that I implemented. With the lookup function patch (that add the q value to the input parameter, the same assigned to you) Chris can invoke the lookup function three time with the three subscriber and obviously three different qvalue. After

[OpenSIPS-Users] R: DB version error in upgrading to 1.5

Hi Carlo, you need to modify also the table_version value in the version table. Regards, MD Ciao Carlo, c'è una tabella in opensips chiamata "version" nel tuo caso devi impostare il valore del campo table_version a 5 in corrispondenza della riga table_name="trusted". Buon Lavoro

[OpenSIPS-Users] call_control module and opensips

Hi All, I'm trying to understand how the call control module works. In my test case, I have two user agents connected to the opensips server. After that a call was estrablished between the two client. I kill the opensips server process and I restart it. When one of the two client hang u

[OpenSIPS-Users] R: Wesip and Opensips

other way to do this?? Regards MD Da: Ginés Gómez [mailto:gi...@voztele.com] Inviato: venerdì 20 marzo 2009 12:46 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: [OpenSIPS-Users] Wesip and Opensips Ciao Mauro, first, you can

[OpenSIPS-Users] Wesip and Opensips

Hi All, I'm setting up the Click2Dial wesip application, and I'm trying to use it. When I start the call in the opensips script I need to manipulate it. So in the local_route branch I try to see if the To party is an alias but I can't use alias_db_lookup function in the local_route bran

[OpenSIPS-Users] Opensips restart question

Hi All, If during a dialog call (after the established phase) I kill the opensips server and after I restart it. If the server receive a BYE message it will route the message correctly, but the call_control module isn't invoked... Is there a solution? My question is, if I use a cluster

[OpenSIPS-Users] Media Proxy on hold timeout

Hi All, I set the on hold timeout to 10 seconds, I do a call with to sip client. After a dialog is established I kill one of the two client, but after 10 seconds the opensips server never receive the dlg_end_dlg command. Can onybody tell me what I'm wrong? Thanks a lot MD

[OpenSIPS-Users] R: R: R: R: R: R: R: R: serialize_branches() and q value....

Hi Bogdan Well done!!! Now it works fine!!! Thank You MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: martedì 10 marzo 2009 21:42 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: R: R: R: R: R: [OpenSIPS-Users]

[OpenSIPS-Users] R: R: R: R: R: R: R: serialize_branches() and q value....

Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: lunedì 9 marzo 2009 13:40 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: R: R: R: R: [OpenSIPS-Users] R: R: serialize_branches() and q value Hi Mauro, indeed there is a issue here. T

[OpenSIPS-Users] R: R: R: R: R: R: R: serialize_branches() and q value....

Now I try to update the source code. Thank you very much MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: lunedì 9 marzo 2009 13:40 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: R: R: R: R: [OpenSIPS-Users]

[OpenSIPS-Users] R: MediaProxy call termination

Thanks Dan, this solution works only for opensips 1.5. Is it correct? Thanks a lot MD -Messaggio originale- Da: Dan Pascu [mailto:d...@ag-projects.com] Inviato: giovedì 5 marzo 2009 22:14 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: [OpenSIPS-Users] MediaProxy

[OpenSIPS-Users] MediaProxy call termination

Hi All, I have a question. If the RTP stream don't pass trough the Media proxy after a timeout the call is closed. Tha call control module receive the stop event correctly, but in the account table isn't present the BYE message (the call is closed by the Media Proxy). There is a way to

[OpenSIPS-Users] R: R: R: R: R: R: serialize_branches() and q value....

value associated to a sip uri or it is possible only in registration phase? Thanks. -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: giovedì 5 marzo 2009 10:25 A: Mauro Davi'; users@lists.opensips.org Oggetto: Re: R: R: R: [OpenSIPS-Users] R: R: seri

[OpenSIPS-Users] R: R: R: R: serialize_branches() and q value....

Hi Bogdan, yes I did in this way... But the problem, I think, is that the qvalue aren't loaded correctly from the DB. Regards, MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: martedì 3 marzo 2009 18:06 A: Mauro Davi'

[OpenSIPS-Users] R: R: R: serialize_branches() and q value....

n't inserted correctly... Is it a bug? Thanks in advance MD -Messaggio originale- Da: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] Per conto di Bogdan-Andrei Iancu Inviato: giovedì 26 febbraio 2009 19:08 A: Mauro Davi' Cc: users@lists.opensip

[OpenSIPS-Users] R: R: serialize_branches() and q value....

ou tell me how I can solve this problem? There is a work-around? Or is there a correct way to call a group of user associated to a SIP URI via serial forking? Thank in andvance MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: mercoledì 25 febbr

[OpenSIPS-Users] R: serialize_branches() and q value....

e log result is ever: "DBG:core:serialize_branches: nothing to do - all same q!" Coould you help me, please? Thanks a lot MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: mercoledì 25 febbraio 2009 14:54 A: Mauro Davi' Cc: users@lists.opensi

[OpenSIPS-Users] serialize_branches() and q value....

Hi All, I saw the documentation but it is to much difficult for me :-) (really I think that the folloeing information is missed...), so I have a question. The below script code add two new destination to my voip platform, paral...@domain.com and ser...@domain.com. The first one works g

[OpenSIPS-Users] Lawful intercept

Hi All, I need to introduce in my architecture a network element that can permit the lawful interceptions. I see that there is an AG Projects MSRP Relay. The goal is to send the mixed received RTP Audio stream from the two parties to an another PSTN gateway... Is there somebody that had res

[OpenSIPS-Users] R: Missing call control module!

I compile the call_control module against the 1.4.2 and 1.4.4 opensips version, and seems to work... I create a server application that communicate with the call_controll module and I receive the INIT, START and STOP messages. Is it only an impression? -Messaggio originale- Da: users-bo

[OpenSIPS-Users] R: R: R: Milliseconds in the accounting table

Thanks so much Bogdan, now the script works fine. -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: martedì 17 febbraio 2009 13:38 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: R: R: [OpenSIPS-Users] Milliseconds in the accounting tabl

[OpenSIPS-Users] R: R: Milliseconds in the accounting table

uot;200 OK received $avp(i:901) $avp(i:902)!!!\n"); } } Is there something wrong? I need to use var(x) or can I use the avp variable? Regards MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: martedì 17 febbraio 2009 00:02 A: Mauro Davi

[OpenSIPS-Users] R: Milliseconds in the accounting table

: sabato 14 febbraio 2009 22:22 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: [OpenSIPS-Users] Milliseconds in the accounting table Hi Mauro, The "time" for accounting is when the reply is received - for the acc'ed INVITEs, it is the time for the 200 OK reply. So, w

[OpenSIPS-Users] Milliseconds in the accounting table

Hi, I'm always a newbye so be patient. I need to trace in the accounting table the start/stop dialog time in milliseconds. I don't know if this is the correct way, but I modified the cfgutils. Now I can write $time(msec) and I receive the millisecs... So I store this information in an avp v

[OpenSIPS-Users] Ending a dialog

Hi All, I'm setting up an opensips architecture with several opensips server. Some of this acts as Proxy same as UAS same as registrar. Every server works on a set of dialogs, if I want end a dialog simply sent to it an XML/RPC command with the appropriate identifier and two BYE are sent (one

[OpenSIPS-Users] R: Can anyone help me?!?

he TM module... Best regards and Thanks so much. MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: martedì 10 febbraio 2009 12:58 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: [OpenSIPS-Users] Can anyone help me?!? Hello M