[OpenSIPS-Users] MWI light on cisco phones
Hi All, I have been trying for a while now I have the mail box registered using active watchers table. I have presence and presence_mwi modules installed. I use sipsak to send a notify packet from the media server (voicemail server asterisk) over to the opensips proxy, I can see from sip traces and TCP dump that the notify packet reaches the phone and the 200 ok comes back but it doesn’t turn the MWI light on or off. Any ideas where I might be going wrong ? I am also using the following in my sip file: CSeq: 201 NOTIFY Event: message-summary Content-Type: application/simple-message-summary Content-Length: 0 Messages-Waiting: no Voicemail: 0/0 Regards, Brian Southworth Communications Developer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] On hold
I don’t think im trying to re reroute re invites not that ive noticed would it help if you took a look on my cfg I am still new to opensips and learning as I go along. But I will also take another look thanks. Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 26 February 2018 16:21 To: users@lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold You need to make sure that in your config you are not trying to re-route in dialog requests (like reINVITE). In dialog requests are routed according to the loose routing mechanism. -ovidiu On Feb 26, 2018 11:01, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Seems to be an issue with it sending the on hold invite to get the call back to the wrong place on inbound calls. Example caller calls in --- agent 1 picks up --- agent 1 places them on hold all is fine --- agent 1 takes call off hold call is dropped Seems that the invite to get the call back gets sent to .192 when it should go back to .193 This doesn’t happen is I use rewritehostport but then on hold calls only work for calls coming from that media server t_relay works across the board but doesn’t actually work for inbound. Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org <mailto:users-boun...@lists.opensips.org> ] On Behalf Of Ovidiu Sas Sent: 26 February 2018 15:49 To: users@lists.opensips.org <mailto:users@lists.opensips.org> users mailto:users@lists.opensips.org> > Subject: Re: [OpenSIPS-Users] On hold You need to take a look at the signalling and figure out who is dropping the call and why. -ovidiu On Feb 26, 2018 10:45, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Please ignore my last email. Both ways work it was an error in my code I wasn’t using t_relay(); Thanks for your help 😊 Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org <mailto:users-boun...@lists.opensips.org> ] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users@lists.opensips.org <mailto:users@lists.opensips.org> users mailto:users@lists.opensips.org> > Subject: Re: [OpenSIPS-Users] On hold Hello Brian, You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 <http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408> Try to use it and see if this will solve your issue. Regards, Ovidiu Sas On Feb 23, 2018 4:13 AM, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips
Re: [OpenSIPS-Users] On hold
Seems to be an issue with it sending the on hold invite to get the call back to the wrong place on inbound calls. Example caller calls in --- agent 1 picks up --- agent 1 places them on hold all is fine --- agent 1 takes call off hold call is dropped Seems that the invite to get the call back gets sent to .192 when it should go back to .193 This doesn’t happen is I use rewritehostport but then on hold calls only work for calls coming from that media server t_relay works across the board but doesn’t actually work for inbound. Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 26 February 2018 15:49 To: users@lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold You need to take a look at the signalling and figure out who is dropping the call and why. -ovidiu On Feb 26, 2018 10:45, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Please ignore my last email. Both ways work it was an error in my code I wasn’t using t_relay(); Thanks for your help 😊 Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org <mailto:users-boun...@lists.opensips.org> ] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users@lists.opensips.org <mailto:users@lists.opensips.org> users mailto:users@lists.opensips.org> > Subject: Re: [OpenSIPS-Users] On hold Hello Brian, You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 <http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408> Try to use it and see if this will solve your issue. Regards, Ovidiu Sas On Feb 23, 2018 4:13 AM, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] On hold
Please ignore my last email. Both ways work it was an error in my code I wasn’t using t_relay(); Thanks for your help 😊 Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users@lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold Hello Brian, You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 <http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408> Try to use it and see if this will solve your issue. Regards, Ovidiu Sas On Feb 23, 2018 4:13 AM, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] On hold
Hi Ovidiu, I tried this in a if statement it works but same thing happens I go to take the call off hold and it just drops the call. I am also using v2.2 but I did check those docs too. Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users@lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold Hello Brian, You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 <http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408> Try to use it and see if this will solve your issue. Regards, Ovidiu Sas On Feb 23, 2018 4:13 AM, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] On hold
Hi Ovidiu, I tried this in a if statement it works but same thing happens I go to take the call off hold and it just drops the call. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users@lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold Hello Brian, You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 <http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408> Try to use it and see if this will solve your issue. Regards, Ovidiu Sas On Feb 23, 2018 4:13 AM, "Brian Southworth" mailto:brian.southwo...@clocom.uk> > wrote: Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] On hold
Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] migrating to opensips 2.3 from 2.2
Hi All, I seem to be getting an error when running the migration. It is also saying the following when I run the following command: opensipsdbctl migrate opensips_2_2 opensips_2_3 INFO: MySQL DB migration tool for OpenSIPS 2.1.x databases -- WARNING: We recommend using this tool ONLY in order to upgrade an existing OpenSIPS 2.1.x MySQL database to the 2.2 schema. Behaviour when automatically migrating earlier DB versions (1.8, 1.9, 1.10, 1.11) to 2.2 is undefined my version is 2.2.3 also when I enter my password for mysql (this is an external DB not internal) I get this ERROR 1045 (28000): Access denied for user 'opensips'@'local_ip.lightspeed.jcsnms.sbcglobal.net' (using password: YES) ERROR 1045 (28000): Access denied for user 'opensips'@'local_ip.lightspeed.jcsnms.sbcglobal.net' (using password: YES) Any idea how I fix this ? I want to start trying to use the EBR module which was made available in opensips 2.3 Regards, Brian Southworth Communications Developer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Database search inside opensips
Hi Opensips, Sorry for all my emails lately. Is there any way I can run a database search in opensips config while a route is being processed. Example scenario: Company has over 15 clients each with their own user id (lets call this a callgroup ID), so the call group id needs to be changed based on the extension calling to match the correct call group. $rU->accountcode (this is just an example). I just thought being able to change this on the fly would be easier than writing loads of new call groups into the Even based routing. Regards, Brian Southworth Communications Developer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Bogan, Thanks for the reply, so are you saying the load balancer will send the call over to the B2B and then to asterisk ? Again sorry for my lack of knowledge there is still a lot I don’t understand or know. Regards, Brian Southworth Communications Developer From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: 12 February 2018 14:23 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] Hi Brian, In this case, I guess that the OPenSIPS B2B (handling the REFER) should sit between the OpenSIPS LB and Asterisk - again , this is the case only if the result of the transfer is a call to an Asterisk box too. If the call may be redirected back to a carrier, the OpenSIPS B2B should sit in front of the LB. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/07/2018 03:38 PM, Brian Southworth wrote: Opensips handles the refer sending it to the asterisk box Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org <mailto:bog...@opensips.org> ] Sent: 07 February 2018 11:36 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <mailto:users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] So the target of the refer is to another Asterisk or may be also back to the carrier ? Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/07/2018 01:32 PM, Brian Southworth wrote: Hi Bogdan, The Cisco phone, generates the refer once you press the xfer button when inside a call. Caller àopensipsà asteriskàCarrier (cisco) Regards, Brian Southworth T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org <mailto:bog...@opensips.org> ] Sent: 07 February 2018 09:38 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <mailto:users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] Hi Brian, Which party is generating the REFER ? the asterisk boxes from behind the LB ? or the carrier side ? and yes, see you in Amsterdam !! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/05/2018 05:52 PM, Brian Southworth wrote: I think I get it now thank you Bogdan. So I would forward the traffic using the opensips proxy, using the if (is_method(“refer”)) to the opensips box that would be the B2BUA? To bridge the call ?. Also look forward to Opensips summit in may 😊 ill see you all there got it booked Saturday 😊 Regards, Brian Southworth ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Error
dlg_validate_dialog: Script error - validate function before having a dialog I am trying to fix contact header on the called, ive fixed the register users used to register with a private ip and not local. Now when inbound calls happen they don’t get them due to the call being send to a local ip that isn’t local. So ive used the validate dialog code chunk and I get back the above Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay
Hi Dragomir, Sorry for my lack of knowledge I am still learning as I go (which is why ill be attending summit this year 😊 ) So If i install media proxy this should relay the DTMF tones ? would I need to install the media proxy on asterisk box and the opensips proxy ? Again sorry for all the questions some of this is still very new to me. Many thanks. Regards, Brian Southworth From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir Haralambiev Sent: 08 February 2018 15:21 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay Hi, Opensips is signaling proxy. You need media proxy to make this. Regards, Dragomir 2018-02-08 17:10 GMT+02:00 Brian Southworth mailto:brian.southwo...@clocom.uk> >: Hi All, How would I go about getting a opensips as a proxy to recognise the DTMF tones or forward them onto asterisk ? Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips to recognise and send DTMF or relay
Hi All, How would I go about getting a opensips as a proxy to recognise the DTMF tones or forward them onto asterisk ? Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Bogdan, The Cisco phone, generates the refer once you press the xfer button when inside a call. Caller opensips asteriskCarrier (cisco) Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: 07 February 2018 09:38 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] Hi Brian, Which party is generating the REFER ? the asterisk boxes from behind the LB ? or the carrier side ? and yes, see you in Amsterdam !! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.openutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/05/2018 05:52 PM, Brian Southworth wrote: I think I get it now thank you Bogdan. So I would forward the traffic using the opensips proxy, using the if (is_method(refer)) to the opensips box that would be the B2BUA? To bridge the call ?. Also look forward to Opensips summit in may 😊 ill see you all there got it booked Saturday 😊 Regards, Brian Southworth From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org <mailto:bog...@opensips.org> ] Sent: 05 February 2018 15:47 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <mailto:users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] Hi Brian, Keep in mind that you cannot make opensips act in the same time as proxy (as required by the load balancer) and as a end-point (as required by the B2BUA). Ideally is to run the two services (LB and B2B) on two opensips instances in a chain. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/02/2018 07:03 PM, Brian Southworth wrote: Sorry my apologies. So from the beginning opensips acts as an authorization proxy which passes the call on to an asterisk box based on load (using load balancer). I am trying to get the opensips proxy to handle call transfers and I thought the b2bua would be the best way. Initially the refer was sent to the asterisk box. On inbound calls The call comes in from the carrier goes to asterisk, asterisk then passes the sip invite to the proxy which then rings the sip phone. What I wish to achieve is a way to transfer an inbound call to an internal extension or external number. Example: Caller A receives call caller A places call on hold and dials caller B caller B picks up caller A presses cisco xfer and call is passed to caller B I was hoping to achieve this using the proxy or asterisk box if possible. I hope this helps. Regards, Brian Southworth From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org <mailto:bog...@opensips.org> ] Sent: 02 February 2018 16:50 To: Brian Southworth <mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list <mailto:users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] I'm a bit confused. The original report was on a record_route() / loose_route() matter. But you say you have opensips as B2B, so the RR mechanism must not be used in such a case - you act either as a end-point
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
I think I get it now thank you Bogdan.
So I would forward the traffic using the opensips proxy, using the if
(is_method(“refer”)) to the opensips box that would be the B2BUA? To bridge the
call ?.
Also look forward to Opensips summit in may 😊 ill see you all there got it
booked Saturday 😊
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 05 February 2018 15:47
To: Brian Southworth ; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
Keep in mind that you cannot make opensips act in the same time as proxy (as
required by the load balancer) and as a end-point (as required by the B2BUA).
Ideally is to run the two services (LB and B2B) on two opensips instances in a
chain.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<http://www.opensips.org/events/Summit-2018Amsterdam>
On 02/02/2018 07:03 PM, Brian Southworth wrote:
Sorry my apologies.
So from the beginning opensips acts as an authorization proxy which passes the
call on to an asterisk box based on load (using load balancer).
I am trying to get the opensips proxy to handle call transfers and I thought
the b2bua would be the best way. Initially the refer was sent to the asterisk
box.
On inbound calls
The call comes in from the carrier goes to asterisk, asterisk then passes the
sip invite to the proxy which then rings the sip phone.
What I wish to achieve is a way to transfer an inbound call to an internal
extension or external number.
Example:
Caller A receives call à caller A places call on hold and dials caller B à
caller B picks up à caller A presses cisco xfer and call is passed to caller B
I was hoping to achieve this using the proxy or asterisk box if possible.
I hope this helps.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 16:50
To: Brian Southworth
<mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list
<mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
I'm a bit confused. The original report was on a record_route() / loose_route()
matter. But you say you have opensips as B2B, so the RR mechanism must not be
used in such a case - you act either as a end-point, either as a proxy - you
cannot be both for the same call.
Now you have this b2b error, during a call transfer scenario. and you mentioned
LB also :)...so I'm a bit confused - could please try to put all these pieces
together, so I can understand what you are doing ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<http://www.opensips.org/events/Summit-2018Amsterdam>
On 02/02/2018 04:27 PM, Brian Southworth wrote:
Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer
and bridge the calls.
I have the B2bUA working now. However my issue is that its not able to send the
replies.
incoming reply
b2b_reply (B2B.222.7591351.1517580641)
Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply
when a final 200 was sent out
Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply
with tm
Feb 2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply
failed - 408, [B2B.452.342.1517580641]
Do you need anything else to help me debug this ? I am not sure why its failing
to pass the reply with tm, I have enabled the param:
modparam("tm", "pass_provisional_replies", 1)
I should also note that I am using the load balancer module also. This normally
deals with all call distribution. In and out.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 14:20
To: Brian Southworth
<mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list
<mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
Maybe that warning points to a routing error that prevents the REFER to be
route to carrier - make a sip capture to be sure the REFER from A is properly
routed and accepted by the carrier.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
OpenSI
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Sorry my apologies.
So from the beginning opensips acts as an authorization proxy which passes the
call on to an asterisk box based on load (using load balancer).
I am trying to get the opensips proxy to handle call transfers and I thought
the b2bua would be the best way. Initially the refer was sent to the asterisk
box.
On inbound calls
The call comes in from the carrier goes to asterisk, asterisk then passes the
sip invite to the proxy which then rings the sip phone.
What I wish to achieve is a way to transfer an inbound call to an internal
extension or external number.
Example:
Caller A receives call à caller A places call on hold and dials caller B à
caller B picks up à caller A presses cisco xfer and call is passed to caller B
I was hoping to achieve this using the proxy or asterisk box if possible.
I hope this helps.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 02 February 2018 16:50
To: Brian Southworth ; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
I'm a bit confused. The original report was on a record_route() / loose_route()
matter. But you say you have opensips as B2B, so the RR mechanism must not be
used in such a case - you act either as a end-point, either as a proxy - you
cannot be both for the same call.
Now you have this b2b error, during a call transfer scenario. and you mentioned
LB also :)...so I'm a bit confused - could please try to put all these pieces
together, so I can understand what you are doing ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<http://www.opensips.org/events/Summit-2018Amsterdam>
On 02/02/2018 04:27 PM, Brian Southworth wrote:
Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer
and bridge the calls.
I have the B2bUA working now. However my issue is that its not able to send the
replies.
incoming reply
b2b_reply (B2B.222.7591351.1517580641)
Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply
when a final 200 was sent out
Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply
with tm
Feb 2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply
failed - 408, [B2B.452.342.1517580641]
Do you need anything else to help me debug this ? I am not sure why its failing
to pass the reply with tm, I have enabled the param:
modparam("tm", "pass_provisional_replies", 1)
I should also note that I am using the load balancer module also. This normally
deals with all call distribution. In and out.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 14:20
To: Brian Southworth
<mailto:brian.southwo...@clocom.uk> ; OpenSIPS users mailling list
<mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
Maybe that warning points to a routing error that prevents the REFER to be
route to carrier - make a sip capture to be sure the REFER from A is properly
routed and accepted by the carrier.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<http://www.opensips.org/events/Summit-2018Amsterdam>
On 02/02/2018 01:38 PM, Brian Southworth wrote:
Hi Bogdan,
Thank you very much, so this doesn’t have any impact on why the call being
transferred are dropped ?
I am trying to transfer a call using the refer method as that is what the cisco
phones use.
The network is setup like so opensips proxy à asterisk gateway à carrier
Scenario:
Inbound call comes into the phone like so: carrier à ast à proxy à phone A
Phone A needs to transfer call to phone B: Phone A dials phone B à phone B
picks up à phone A presses xfer button and call is dropped.
Any help would be appreciated.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 11:29
To: OpenSIPS users mailling list
<mailto:users@lists.opensips.org> ; Brian Southworth
<mailto:brian.southwo...@clocom.uk>
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
That warning means OpenSIPS found a Route header (while doing loose_route) that
is suppose to be of its own, but the network information from the header
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer
and bridge the calls.
I have the B2bUA working now. However my issue is that its not able to send the
replies.
incoming reply
b2b_reply (B2B.222.7591351.1517580641)
Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply
when a final 200 was sent out
Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply
with tm
Feb 2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply
failed - 408, [B2B.452.342.1517580641]
Do you need anything else to help me debug this ? I am not sure why its failing
to pass the reply with tm, I have enabled the param:
modparam("tm", "pass_provisional_replies", 1)
I should also note that I am using the load balancer module also. This normally
deals with all call distribution. In and out.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 02 February 2018 14:20
To: Brian Southworth ; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
Maybe that warning points to a routing error that prevents the REFER to be
route to carrier - make a sip capture to be sure the REFER from A is properly
routed and accepted by the carrier.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<http://www.opensips.org/events/Summit-2018Amsterdam>
On 02/02/2018 01:38 PM, Brian Southworth wrote:
Hi Bogdan,
Thank you very much, so this doesn’t have any impact on why the call being
transferred are dropped ?
I am trying to transfer a call using the refer method as that is what the cisco
phones use.
The network is setup like so opensips proxy à asterisk gateway à carrier
Scenario:
Inbound call comes into the phone like so: carrier à ast à proxy à phone A
Phone A needs to transfer call to phone B: Phone A dials phone B à phone B
picks up à phone A presses xfer button and call is dropped.
Any help would be appreciated.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org
<mailto:bog...@opensips.org> ]
Sent: 02 February 2018 11:29
To: OpenSIPS users mailling list
<mailto:users@lists.opensips.org> ; Brian Southworth
<mailto:brian.southwo...@clocom.uk>
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Brian,
That warning means OpenSIPS found a Route header (while doing loose_route) that
is suppose to be of its own, but the network information from the header does
not match any of the OpenSIPS SIP listeners.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
<http://www.opensips.org/events/Summit-2018Amsterdam>
On 02/02/2018 11:14 AM, Brian Southworth wrote:
I get this when trying to transfer calls using the B2BUA:
[15066] WARNING:rr:after_strict: no socket found to match RR
[1][xxx.xxx.xxx.xxx:5060]
When I try looking on the mailing list there are no other similar posts, could
you please shed some light on what maybe causing this please.
Regards,
Brian Southworth
___
Users mailing list
Users@lists.opensips.org <mailto:Users@lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi Bogdan, Thank you very much, so this doesn’t have any impact on why the call being transferred are dropped ? I am trying to transfer a call using the refer method as that is what the cisco phones use. The network is setup like so opensips proxy à asterisk gateway à carrier Scenario: Inbound call comes into the phone like so: carrier à ast à proxy à phone A Phone A needs to transfer call to phone B: Phone A dials phone B à phone B picks up à phone A presses xfer button and call is dropped. Any help would be appreciated. Regards, Brian Southworth From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: 02 February 2018 11:29 To: OpenSIPS users mailling list ; Brian Southworth Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/02/2018 11:14 AM, Brian Southworth wrote: I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060] When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please. Regards, Brian Southworth ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]
I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060] When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please. Regards, Brian Southworth ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ACK bug help needed
Hi Razvan,
I just checked my cfg.
It contains this:
if (has_totag()) {
if (topology_hiding_match()) {
xlog("Succesfully matched this request to a topology hiding dialog.
\n");
xlog("Calller side callid is $ci \n");
xlog("Callee side callid is $TH_callee_callid \n");
t_relay();
exit;
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else
exit;
}
sl_send_reply("404","Not here");
exit;
}
}
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>
<http://www.facebook.com/clocom.uk>
Like us on Facebook
Follow us on Twitter
Clocom is a green company. Think, do you need to print this email?
This message contains confidential information and is intended only for the
individual named. If you are not the named addressee you should not
disseminate, distribute or copy this e-mail. Please notify the sender
immediately by e-mail if you have received this e-mail by mistake and delete
this e-mail from your system. E-mail transmission cannot be guaranteed to be
secure or error-free as information could be intercepted, corrupted, lost,
destroyed, arrive late or incomplete, or contain viruses. The sender therefore
does not accept liability for any errors or omissions in the contents of this
message, which arise as a result of e-mail transmission. If verification is
required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road,
Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/>
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan
Crainea
Sent: 05 September 2017 10:57
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] ACK bug help needed
Hi, Brian!
OpenSIPS doesn't need to send an ACK back, it has to relay the 200 OK to the
caller, and the caller will send the ACK back. Can you confirm you are doing
record_route() on the initial INVITE?
Also, can you post somewhere a SIP trace/pcap?
Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 09/05/2017 12:44 PM, Brian Southworth wrote:
Hi All,
I seem to be having issues with outbound calls, the calls go out and the
connection is established.
But when the asterisk gateway send the 200OK back from the provider to opensips
proxy, the proxy doesn’t send the ACK packet back to asterisk it just keeps
looping it to itself.
Any help would be appreciated.
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>
<http://www.facebook.com/clocom.uk>
Like us on Facebook
Follow us on Twitter
Clocom is a green company. Think, do you need to print this email?
This message contains confidential information and is intended only for the
individual named. If you are not the named addressee you should not
disseminate, distribute or copy this e-mail. Please notify the sender
immediately by e-mail if you have received this e-mail by mistake and delete
this e-mail from your system. E-mail transmission cannot be guaranteed to be
secure or error-free as information could be intercepted, corrupted, lost,
destroyed, arrive late or incomplete, or contain viruses. The sender therefore
does not accept liability for any errors or omissions in the contents of this
message, which arise as a result of e-mail transmission. If verification is
required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road,
Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/>
___
Users mailing list
Users@lists.opensips.org <mailto:Users@lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
___
Users mailing list
Users@lists.opensips.org <mailto:Users@lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ACK bug help needed
Hi Razvan, I Have sent you the pcap file link to your personal email. Asterisk cuts the calls because it is missing a critical packet ACK from what I can see replying to the 200ok it sent to opensips. But opensips only sends the ACK packet to itself and nowhere else, the call has audio and is active for 33 seconds until the call is cut. Due to a no reply to critical packet 101. The system works by using opensips as a proxy open sips then sets the dst uri and the call is then t_relay() on to asterisk gateway. Yeah I can confirm I am doing Record_route(); Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: 05 September 2017 10:57 To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] ACK bug help needed Hi, Brian! OpenSIPS doesn't need to send an ACK back, it has to relay the 200 OK to the caller, and the caller will send the ACK back. Can you confirm you are doing record_route() on the initial INVITE? Also, can you post somewhere a SIP trace/pcap? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com <http://www.opensips-solutions.com> On 09/05/2017 12:44 PM, Brian Southworth wrote: Hi All, I seem to be having issues with outbound calls, the calls go out and the connection is established. But when the asterisk gateway send the 200OK back from the provider to opensips proxy, the proxy doesn’t send the ACK packet back to asterisk it just keeps looping it to itself. Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ACK bug help needed
Hi All, I seem to be having issues with outbound calls, the calls go out and the connection is established. But when the asterisk gateway send the 200OK back from the provider to opensips proxy, the proxy doesn’t send the ACK packet back to asterisk it just keeps looping it to itself. Any help would be appreciated. Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/clocom.uk> Like us on Facebook Follow us on Twitter Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Fw: Outbound Call issue
10:10:45] [Jun 21 10:10:45] <--- Reliably Transmitting (NAT) to opensips:5060 ---> [Jun 21 10:10:45] SIP/2.0 487 Request Terminated [Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0;received=34.250.75.163;rport=5060 [Jun 21 10:10:45] Via: SIP/2.0/UDP office IP:5060;branch=z9hG4bK-af4637c [Jun 21 10:10:45] From: "opensips" ;tag=6cf20f07e3486c44o3 [Jun 21 10:10:45] To: "Brian07476243394" ;tag=as29b37eb3 [Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8@192.168.1.48 [Jun 21 10:10:45] CSeq: 101 INVITE [Jun 21 10:10:45] Server: Asterisk PBX GIT-master-b05d2fd [Jun 21 10:10:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 21 10:10:45] Supported: replaces, timer [Jun 21 10:10:45] Content-Length: 0 [Jun 21 10:10:45] [Jun 21 10:10:45] [Jun 21 10:10:45] <> [Jun 21 10:10:45] [Jun 21 10:10:45] <--- Transmitting (NAT) to opensips:5060 ---> [Jun 21 10:10:45] SIP/2.0 200 OK [Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0;received=opensips;rport=5060 [Jun 21 10:10:45] From: "opensips" ;tag=6cf20f07e3486c44o3 [Jun 21 10:10:45] To: "Brian07476243394" ;tag=as29b37eb3 [Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8@192.168.1.48 [Jun 21 10:10:45] CSeq: 101 CANCEL [Jun 21 10:10:45] Server: Asterisk PBX GIT-master-b05d2fd [Jun 21 10:10:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 21 10:10:45] Supported: replaces, timer [Jun 21 10:10:45] Content-Length: 0 [Jun 21 10:10:45] [Jun 21 10:10:45] [Jun 21 10:10:45] <> [Jun 21 10:10:45] [Jun 21 10:10:45] <--- SIP read from UDP:opensips:5060 ---> [Jun 21 10:10:45] ACK sip:07476243394@opensips SIP/2.0 [Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0 [Jun 21 10:10:45] From: "opensips" ;tag=6cf20f07e3486c44o3 [Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8@192.168.1.48 [Jun 21 10:10:45] To: "Brian07476243394" ;tag=as29b37eb3 [Jun 21 10:10:45] CSeq: 101 ACK [Jun 21 10:10:45] Max-Forwards: 70 [Jun 21 10:10:45] User-Agent: OpenSIPS (2.2.3 (x86_64/linux)) [Jun 21 10:10:45] Content-Length: 0 [Jun 21 10:10:45] [Jun 21 10:10:45] <-> I for security reasons I have edited any IP’s if you need the unedited version I will send it in an email directly Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 DDI:01625 837112 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/nima.it.solutions> Like us on Facebook Follow us on Twitter <http://www.twitter.com/NIMA_IT> Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] FW: Problem with outbound calls
Hi All, I seem to have an issue when using forward() my cisco phone says loop detected. If I used t_relay() my phone will say invalid destination both result in a bye signal being sent by open sips, however the call is still passed to the carrier Regards, Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 0 446677 W: www.clocom.uk <http://www.clocom.uk/> <http://www.facebook.com/nima.it.solutions> Like us on Facebook Follow us on Twitter <http://www.twitter.com/NIMA_IT> Clocom is a green company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk <http://www.clocom.uk/> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
