[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

[OpenSIPS-Users] Fw: Outbound Call issue

94" <sip:07476243394@opensips>;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]

Re: [OpenSIPS-Users] ACK bug help needed

    } else     exit;     }     sl_send_reply("404","Not here");     exit;     }     }   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3E

Re: [OpenSIPS-Users] ACK bug help needed

olutions.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

[OpenSIPS-Users] ACK bug help needed

.   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>

Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

and out.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: 02 February 2018 14:20 To: Brian Southworth <brian.southwo...@clocom.uk>; OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_s

Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

.   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

Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

  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

[OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

.   Regards,   Brian Southworth ___ 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]

   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: 05 February 2018 15:47 To: Brian Southworth <brian.southwo...@clocom.uk>; OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_stric

[OpenSIPS-Users] Error

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

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

[OpenSIPS-Users] Database search inside opensips

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 Communicati

[OpenSIPS-Users] migrating to opensips 2.3 from 2.2

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

[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

Re: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay

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 <users@lists.opensips.org> Subject: Re: [Op

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 <h

[OpenSIPS-Users] On hold

e 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/>      

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

[OpenSIPS-Users] MWI light on cisco phones

-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

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

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

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 <h

Re: [OpenSIPS-Users] On hold

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