[asterisk-users] call recording via 3rd INVITE/SIP leg
I have a call recording (audio) requirement that isn't addressed by local Monitor/Record features. All signalling and media currently pass through the Asterisk servers, so that won't be an issue. Instead of locally recording audio, for certain calls I need to add what is effectively a 3rd leg to the in progress 2-leg call. This 3rd leg is a SIP dial to a URI and/or PSTN number. I'm thinking I have to do this with a conference bridge config and add a 3rd muted leg to the conference? Suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily
I think it's 'divert.noanswer', found in site.cfg, or at least that's where I have it. It's set to enabled and it still doesn't work. Out of curiosity, do you have reg.1.fwd.noanswer.status set anywhere? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz Sent: Wednesday, December 12, 2012 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily I have Polycom IP550. The Forward No Answer is working fine when enabled. I was looking at the sip.cfg but don't know exactly what to look for, can you give me a hint to where would i find that option? Thanks, On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill justin.sherr...@americanrocksalt.commailto:justin.sherr...@americanrocksalt.com wrote: I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for Always works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to No Answer means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a 302 SIP message issued. I can't find anything in the phone settings that look like it would disable this. Anyone else with a Polycom set that sees this, or does not see this and has forward no answer working? Justin Sherrill - American Rock Salt P: 585-991-6825tel:585-991-6825 F: 585-991-6925tel:585-991-6925 C: 585-298-6826tel:585-298-6826 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call recording via 3rd INVITE/SIP leg
Tom Browning wrote: I have a call recording (audio) requirement that isn't addressed by local Monitor/Record features. All signalling and media currently pass through the Asterisk servers, so that won't be an issue. Instead of locally recording audio, for certain calls I need to add what is effectively a 3rd leg to the in progress 2-leg call. This 3rd leg is a SIP dial to a URI and/or PSTN number. I'm thinking I have to do this with a conference bridge config and add a 3rd muted leg to the conference? If you don't want to incur the overhead of a full blown conference bridge you can use ChanSpy to spy on a channel. It will provide a mixed stream of the incoming and outgoing part of the channel. So essentially use Originate to call your 3rd leg and then have it execute ChanSpy with the correct criteria to get to the right leg. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
I am trying to get a digital accoustics talkmaster to register to asterisk 1.4.43 I am getting the 401 unauthorized. I have host=dynamic I have verified the passwords match What else is there? I dont see any further clues in sip set debug. all it says is using request as basis request What do I try? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
Please post the sip.conf entry with any confidential data xxx'ed out. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 2:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 I am trying to get a digital accoustics talkmaster to register to asterisk 1.4.43 I am getting the 401 unauthorized. I have host=dynamic I have verified the passwords match What else is there? I dont see any further clues in sip set debug. all it says is using request as basis request What do I try? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
[5001] type=friend username=5001 secret=XXX dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=incoming host=dynamic canreinvite=no qualify=no trustrpid=yes sendrpid=no nat=no I did notice one more thing: chan_sip.c:17045 handle_request_register: Registration from '5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195' - No matching peer found Why is there no matching peer I have it defined. I shows in my sip show peers? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 [5001] type=friend username=5001 secret=XXX dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=incoming host=dynamic canreinvite=no qualify=no trustrpid=yes sendrpid=no nat=no I did notice one more thing: chan_sip.c:17045 handle_request_register: Registration from '5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195' - No matching peer found Why is there no matching peer I have it defined. I shows in my sip show peers? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. I have tried both friend and peer. I changed the sendrpid to yes and made no difference either. Still get 401 Unauthorized. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. I have tried both friend and peer. I changed the sendrpid to yes and made no difference either. Still get 401 Unauthorized. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. That worked - it registered. Why would it not register the other way? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. That worked - it registered. Why would it not register the other way? Jerry n It's supposed to work both ways. It depends on how you have it set up on the remote side. It's been two years since I went through the process so it isn't fresh on my brain. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call recording via 3rd INVITE/SIP leg
On Thu, Dec 13, 2012 at 9:45 AM, Joshua Colp jc...@digium.com wrote: If you don't want to incur the overhead of a full blown conference bridge you can use ChanSpy to spy on a channel. It will provide a mixed stream of the incoming and outgoing part of the channel. So essentially use Originate to call your 3rd leg and then have it execute ChanSpy with the correct criteria to get to the right leg. Thanks Joshua, I'll check that out! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip-user status
Hi all, I'm caught up in a struggle between people how can not make up their mind... Half way implementing a asterisk farm and they come up with another feature they've seen in kamaillo. What he showed me was this: three registered sip users, a) one changes his presence status on his softphone, and all see the status change. b) one calls another, and the third person see the status of the other two change to busy. I've seen code/dialplan snippets where you could change your status by dialling a specific extension, on which asterisk will react (and change some variables accordingly), but that is not what i'm looking for. It seems that kamaillo has build-in features to react on sip-simple changes. Can i perform the same trick with asterisk? if so, how? Hans. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
