Re: [OpenSIPS-Users] Managing Origination Routes

Hello Bogdan, I saw the light! And it's OpenSIPS Green! :) Thank you so much for your time, the dialplan script is working perfectly. As for SSet 1 and querying the username, and domain fields of dr_groups individually, I will just give the DR documentation a closer look. I know that you are one

Re: [OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

Ok thanks I will testing also :) Le 07/03/2013 21:05, Nick Altmann a écrit : I'm sorry. Correct variant: $(ai{uri.user}) -- Nick 2013/3/8 Nick Altmann > So, you can do $($ai{uri.user}) :-) -- Nick 2013/3/8 Hubert Mickael mailto:mick...@win

Re: [OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

I'm sorry. Correct variant: $(ai{uri.user}) -- Nick 2013/3/8 Nick Altmann > So, you can do $($ai{uri.user}) :-) > > -- > Nick > > 2013/3/8 Hubert Mickael > >> I want just user in the PAI's URI and $ai is all URI: sip:user@domain >> >> Le 07/03/2013 20:31, Brett Nemeroff a écrit : >> >> Wha

Re: [OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

Thanks a lot I testing that tomorrow bye Le 07/03/2013 18:41, Bogdan-Andrei Iancu a écrit : Hi Mickael, You can simply use a transformation: $(hdr(P-Asserted-Identity){nameaddr.uri}{uri.user}) See http://www.opensips.org/Resources/DocsCoreTran19 Regards, Bogdan-Andrei Iancu OpenSIPS Found

Re: [OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

So, you can do $($ai{uri.user}) :-) -- Nick 2013/3/8 Hubert Mickael > I want just user in the PAI's URI and $ai is all URI: sip:user@domain > > Le 07/03/2013 20:31, Brett Nemeroff a écrit : > > What about $ai ? >> >> -Brett >> >> On Mar 7, 2013, at 11:41 AM, Bogdan-Andrei Iancu >> wrote: >>

Re: [OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

I want just user in the PAI's URI and $ai is all URI: sip:user@domain Le 07/03/2013 20:31, Brett Nemeroff a écrit : What about $ai ? -Brett On Mar 7, 2013, at 11:41 AM, Bogdan-Andrei Iancu wrote: Hi Mickael, You can simply use a transformation: $(hdr(P-Asserted-Identity){nameaddr.uri}{uri

Re: [OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

What about $ai ? -Brett On Mar 7, 2013, at 11:41 AM, Bogdan-Andrei Iancu wrote: > Hi Mickael, > > You can simply use a transformation: > > $(hdr(P-Asserted-Identity){nameaddr.uri}{uri.user}) > > See http://www.opensips.org/Resources/DocsCoreTran19 > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS

Re: [OpenSIPS-Users] Managing Origination Routes

Thank you so much for your help Bogdan!!! I know how to query dr_groups for username and domain (i.e., do_routing()) however, unsure on how to query the table for just the domain. An example like the last would be great! Kindly. Also, the previous example of do_translate("1","$ru/$avp(dr_id)"), $a

[OpenSIPS-Users] Multiple re-registrations in location table

Hello, I am in the process of upgrading our version of OpenSIPS from version 1.6.2 to version 1.8.2 and am facing a bit of a problem. I have noticed that re-registrations sent from a UA are saved as NEW entries in the location table. Previously in OpenSIPS 1.6.2, it would overwrite the previous l

Re: [OpenSIPS-Users] b2b terminate_call!

Are you sure it is not a retransmission ?? Could you post a SIP capture of the entire call ? Regards Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03/07/2013 07:49 PM, Jorge Henrique Pinho wrote: Hi Bogdan! The call is terminated in both clients, t

Re: [OpenSIPS-Users] b2b terminate_call!

Hi Bogdan! The call is terminated in both clients, the issue is that b2b is sending two BYE messages to B instead of one. Is there any way to fix this behavior? Kind regards, Jorge Pinho From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: quinta-feira, 7 de Março de 2013 17:44 To: Open

Re: [OpenSIPS-Users] b2b terminate_call!

Hi Jorge, Why is the order important ( A or B first) ? as you simply want to terminate the whole call :). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03/07/2013 05:35 PM, Jorge Henrique Pinho wrote: Hi, I am using opensips with b2b modu

Re: [OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

Hi Mickael, You can simply use a transformation: $(hdr(P-Asserted-Identity){nameaddr.uri}{uri.user}) See http://www.opensips.org/Resources/DocsCoreTran19 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03/07/2013 06:25 PM, Mickael HUBERT wrot

Re: [OpenSIPS-Users] Managing Origination Routes

ups, forgot that do_routing() accepts only AVP vars, so instead of $var, use $avp , correct! About the sets, super sets, etc... - for each case you use a different approach - like for SSet1 use dr_groups table and for SSet2 use via dialplan. Regards, Bogdan Bogdan-Andrei Iancu OpenSIPS Foun

Re: [OpenSIPS-Users] Managing Origination Routes

Hello Bogdan, It seems that do_routing does not accept "$var(id)": ERROR:drouting:fixup_do_routing: malformed or non AVP $var(groupid) AVP definition Mar 7 11:13:12 [5027] ERROR:core:fix_actions: fixing failed (code=-6) at cfg line 482 Do I just use "$avp(id)" or do I have to assign the value

[OpenSIPS-Users] User in SIP Request's P-Asserted-Identity header URI

I list, I want extract User to P-Asserted-Identity URI, but I cannot find a variable. To user's PPI is $pU, but to PAI ? Thanks in advance Mickael ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/us

Re: [OpenSIPS-Users] b2b terminate_call!

Hi Nick. It could be bad formatted or other errors that I cannot fix, i.e. codec names and attributes that do not comply with the RFC. In the cases in which I can’t fix the message, I need to terminated the call. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On

Re: [OpenSIPS-Users] b2b terminate_call!

Why just not to fix 200 OK? -- Nick 2013/3/7 Jorge Henrique Pinho > Hi, I am using opensips with b2b module with topology hiding. > > I am trying to terminate a call when i receive the 200OK response to a > Re-Invite. To accomplish this i am using the 'terminate_call' function > defined in

[OpenSIPS-Users] b2b terminate_call!

Hi, I am using opensips with b2b module with topology hiding. I am trying to terminate a call when i receive the 200OK response to a Re-Invite. To accomplish this i am using the 'terminate_call' function defined in b2b_logic module. The dialog is establish and an user sends an in dialog Invite to

[OpenSIPS-Users] Handling of Call With SDP in Ack

Hi all, I'have a device (CUCM) that send to my Opensips the Invite Without SDP Trace (if i'm not wrong there nothing strange in this), but it send the SDP in the following ACK. In my config, i call engage_media_proxy on the first invite,is this correct? If the call has been answered there ar

Re: [OpenSIPS-Users] Managing Origination Routes

Hi Nick, if you want to select the routing group based on caller domain, simply do: do_translate("1","$fd/$avp(dr_id)"); So, the input for dialplan is $fd (domain from FROM hdr) and output is in $avp(dr_id) . In DB put : * /match-op/ = 1 (regexp) * /match_exp/ = "^(

Re: [OpenSIPS-Users] OpenSIPS-CP RTPProxy (nathelper)

Alex, Update working great. You are awesome! Thank you, Seth On 3/7/2013 3:10 AM, Alex Ionescu wrote: Hi Seth, I have done the necessary changes and committed on SVN trunk. Please do a fresh checkout from trunk. The changed files are: ../config/tools/system/nathelper/local.inc.php ../web/to

Re: [OpenSIPS-Users] OpenSIPS-CP RTPProxy (nathelper)

Hi Seth, I have done the necessary changes and committed on SVN trunk. Please do a fresh checkout from trunk. The changed files are: ../config/tools/system/nathelper/local.inc.php ../web/tool/system/nathelper/natall.php ../web/tool/system/nathelper/template/natall.main.php Check and let me know