[asterisk-users] PJSIP issues with handling incoming calls
Hello guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005@80.75.132.66 trunk2: 73432260050@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from 'sip:+ 73432260005@80.75.132.66' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set identify by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts?-- _ -- 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] PJSIP issues with handling incoming calls
Le 02/09/2014 08:47, Nick Awesome a écrit : Hello guys. Hi Have 2 external numbers that required registration on provider server, trunk1: 734322600*05*@80.75.132.66 trunk2: 734322600*50*@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from 'sip:+ 734322600*05*@80.75.132.66' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set /identify /by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts? You have to register to the gateway with each account user and password like sip.conf register = 734322600*05*:password1@myProvider/734322600*05* register = 734322600*50*:password2@myProvider/734322600*50* [myProvider] type=peer host=80.75.132.66 context=from-myProvider ... extensions.conf [from-myProvider] exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*) ... exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*) ... -- _ -- 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] AGI scripts - delay issue.
Am 02.09.2014 07:09, schrieb Bryant Zimmerman: Hey All We have several AGI scripts that access databases. These work well most of the time. The issue we are having is that on rare occasion our script must fail to a backup database server. When this occurs it may take up to two seconds to do so. The issue is when there is this delay the script loses access to read global channel variable values only after the delay. This is driving me crazy is there some kind of AGI timeout issue or bug that could be causing this. What do you mean with the script loses access to read global channel variable values? What is the asterisk version? What channel tech is used? What type of AGI-Scripts do you use? -- _ -- 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] PJSIP issues with handling incoming calls
So there is no way to do that with pjsip? On 02 Sep 2014, at 11:35, Administrator TOOTAI ad...@tootai.net wrote: Le 02/09/2014 08:47, Nick Awesome a écrit : Hello guys. Hi Have 2 external numbers that required registration on provider server, trunk1: 734322600*05*@80.75.132.66 trunk2: 734322600*50*@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from 'sip:+ 734322600*05*@80.75.132.66' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set /identify /by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts? You have to register to the gateway with each account user and password like sip.conf register = 734322600*05*:password1@myProvider/734322600*05* register = 734322600*50*:password2@myProvider/734322600*50* [myProvider] type=peer host=80.75.132.66 context=from-myProvider ... extensions.conf [from-myProvider] exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*) ... exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*) ... -- _ -- 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] PJSIP issues with handling incoming calls
Le 02/09/2014 09:38, Nick Awesome a écrit : So there is no way to do that with pjsip? Sorry, I didn't read carefully the subject. I can't answer for pjsip. My bad :-( On 02 Sep 2014, at 11:35, Administrator TOOTAI ad...@tootai.net wrote: Le 02/09/2014 08:47, Nick Awesome a écrit : Hello guys. Hi Have 2 external numbers that required registration on provider server, trunk1: 734322600*05*@80.75.132.66 trunk2: 734322600*50*@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from 'sip:+ 734322600*05*@80.75.132.66' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set /identify /by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts? You have to register to the gateway with each account user and password like sip.conf register = 734322600*05*:password1@myProvider/734322600*05* register = 734322600*50*:password2@myProvider/734322600*50* [myProvider] type=peer host=80.75.132.66 context=from-myProvider ... extensions.conf [from-myProvider] exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*) ... exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*) ... -- _ -- 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
[asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk
Hello, I have a situation where a call comes in to my Asterisk server B. This call comes from another Asterisk server A. I want to tell to this server A why my server B hangs up. So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() But I notice that this extra SIP-header is not send within the SIP-reponse : SIP/2.0 603 Declined Via: SIP/2.0/UDP xx.xx.xx.98:5060;branch=z9hG4bK168884d7;received=xx.xx.xx.98;rport=5060 From: 5006 sip:5...@xx.xx.xx.98;tag=as50c98b4c To: sip:0...@xx.xx.xx.238;tag=as3c6e57b0 Call-ID: 6d1039bb22716c6e6dec69fb3e78a...@xx.xx.xx.98:5060 CSeq: 102 INVITE Server: myasterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 How can I make this work ? Thanks. Jonas. -- _ -- 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] POSSA CID Superfecta to query a CardDAV server ?
On 01/09/14 12:05, Marie Fischer wrote: Well, you made me curious - wrote up a little perl script to do a filtered report by phone number. It takes 2-3 seconds to get a response from OSX server (Mavericks). Which sure is shorter then doing a full sync, but still longish. Would be interesting to know how long other servers take. I can try to time my CardDAV query program (keeping in mind that it runs on the same physical segment and subnet as the server) (sorry I don't know Perl ;) I'm a Free Pascal user ) Will come back to that. Now, for CID you would want this to run in your dial plan after the call comes in and before you Dial() your local extension. One ringtone is 5 seconds (1 sec tone, 4 sec silence), so it's actually not too bad (remember those analog caller ID boxes which got the caller ID between first and second ringtone?). Maybe you'd need to send Progress() or Ringing() back to the calling party. Hmm ok. Let me time it and see the timings. Lukasz -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be wrote: So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() SIPAddHeader only works for INVITE as far as I know. Steve-- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 02-09-14 11:34, Steven Howes wrote: On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() SIPAddHeader only works for INVITE as far as I know. Steve OK. Then how can I let another Asterisk server know the custom reason of hangup ? If it is not possible with custom SIP-header, then how ? Regards, Jonas. -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 2 Sep 2014, at 10:38, Jonas Kellens jonas.kell...@telenet.be wrote: Then how can I let another Asterisk server know the custom reason of hangup ? If it is not possible with custom SIP-header, then how ? As far as I know that’s going to require a source change. May not be the case with PJSIP though - not used that yet. Steve-- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On Tuesday 02 Sep 2014, Jonas Kellens wrote: On 02-09-14 11:34, Steven Howes wrote: On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() SIPAddHeader only works for INVITE as far as I know. Steve OK. Then how can I let another Asterisk server know the custom reason of hangup ? If it is not possible with custom SIP-header, then how ? Fire off an AGI script which will (somehow) send the necessary message to the other Asterisk server. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] PJSIP issues with handling incoming calls
Nick Awesome wrote: Hello guys. Kia ora, Have 2 external numbers that required registration on provider server, trunk1: 734322600*05*@80.75.132.66 trunk2: 734322600*50*@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from 'sip:+ 734322600*05*@80.75.132.66' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set /identify /by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts? If the From header contains the destination number (as it seems to based on your above log message and config) you can create two different endpoints and match based on the user portion of the From header. [734322600*05*] type=endpoint context=did-1 disallow=all allow=ulaw [734322600*50*] type=endpoint context=did-2 disallow=all allow=ulaw If this is not correct then you can only match once based on the source IP address currently. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] PJSIP issues with handling incoming calls
Thats because I call from one to other here’s logs where I call from mobile --- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 --- ACK sip:s@pbx_ip_address:57408;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26;rport Max-Forwards: 70 To: sip:73432260005@80.75.132.66;tag=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26 From: sip:+7823064@80.75.132.66;tag=7ozmpvsvqs26kcor.o Call-ID: 18e2786560719216837824k41099rmwp CSeq: 586 ACK Content-Length: 0 --- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 --- ACK sip:s@pbx_ip_address:57408;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400;rport Max-Forwards: 70 To: sip:73432260050@80.75.132.66;tag=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400 From: sip:+7823064@80.75.132.66;tag=yddmzvcoi3waw24e.o Call-ID: 22e7064301970213226722k41100rmwp CSeq: 588 ACK Content-Length: 0 On 02 Sep 2014, at 15:01, Joshua Colp jc...@digium.com wrote: Nick Awesome wrote: Hello guys. Kia ora, Have 2 external numbers that required registration on provider server, trunk1: 734322600*05*@80.75.132.66 trunk2: 734322600*50*@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from 'sip:+ 734322600*05*@80.75.132.66' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set /identify /by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts? If the From header contains the destination number (as it seems to based on your above log message and config) you can create two different endpoints and match based on the user portion of the From header. [734322600*05*] type=endpoint context=did-1 disallow=all allow=ulaw [734322600*50*] type=endpoint context=did-2 disallow=all allow=ulaw If this is not correct then you can only match once based on the source IP address currently. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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 -- _ -- 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] PJSIP issues with handling incoming calls
Nick Awesome wrote: Thats because I call from one to other Then no, you can only match based on IP address. This also applies to chan_sip. You have to send both to the same context and then within there you can differentiate them based on the dialed number. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] PJSIP issues with handling incoming calls
On Tuesday 02 Sep 2014, Nick Awesome wrote:
Hello guys.
Have 2 external numbers that required registration on provider server,
trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk can’t match endpoint
Request from 'sip:+ 73432260005@80.75.132.66' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found
Can’t set identify by IP because they got the same ip.
Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?
Can't you send them both to the same context initially; but once you are
there, match the outside number (which can be found in ${EXTEN} if it is the
number that was dialled from their end, or ${CALLERID(num)} if it is the
number they are calling from) within that context and use a GoToIf() to send
calls from trunk 2 to the correct context?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
-- 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
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Re: [asterisk-users] PJSIP issues with handling incoming calls
Tried doing that, but
first: AGI-exten is ’s’ for some reason.
and second its not practical, for example if 80.75.132.66 wound like to
register on my * server - it will not work because I already using that IP with
different endpoint
well, its critical trouble for me, coming back to chat_sip :|
On 02 Sep 2014, at 15:32, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Tuesday 02 Sep 2014, Nick Awesome wrote:
Hello guys.
Have 2 external numbers that required registration on provider server,
trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk can’t match endpoint
Request from 'sip:+ 73432260005@80.75.132.66' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found
Can’t set identify by IP because they got the same ip.
Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?
Can't you send them both to the same context initially; but once you are
there, match the outside number (which can be found in ${EXTEN} if it is the
number that was dialled from their end, or ${CALLERID(num)} if it is the
number they are calling from) within that context and use a GoToIf() to send
calls from trunk 2 to the correct context?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
-- 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 --
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Re: [asterisk-users] PJSIP issues with handling incoming calls
Nick Awesome wrote: Tried doing that, but first: AGI-exten is ’s’ for some reason. and second its not practical, for example if 80.75.132.66 wound like to register on my * server - it will not work because I already using that IP with different endpoint well, its critical trouble for me, coming back to chat_sip :| How will you do this in chan_sip? The behavior between the two is the same, despite the configuration being different. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] PJSIP issues with handling incoming calls
register = 73432260005:pass@10001 register = 73432260050:pass@10002 [10001] type=peer host=80.75.132.66 context=dialmap [10002] type=peer host=80.75.132.66 context=dialmap so now in context dialmap (agi application) AGI-agi_channel is 'SIP/10001-0005’ parsing 10001 and checking db for matches, in db I have table with all my trunks information On 02 Sep 2014, at 15:49, Joshua Colp jc...@digium.com wrote: Nick Awesome wrote: Tried doing that, but first: AGI-exten is ’s’ for some reason. and second its not practical, for example if 80.75.132.66 wound like to register on my * server - it will not work because I already using that IP with different endpoint well, its critical trouble for me, coming back to chat_sip :| How will you do this in chan_sip? The behavior between the two is the same, despite the configuration being different. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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 -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
Try Hangup(123) where 123 is whatever hangup cause you want to send back to the caller. The calliing Asterisk server will get the valuse back in HANGUPCAUSE variable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, September 02, 2014 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk Hello, I have a situation where a call comes in to my Asterisk server B. This call comes from another Asterisk server A. I want to tell to this server A why my server B hangs up. So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() But I notice that this extra SIP-header is not send within the SIP-reponse : -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 02-09-14 14:22, Eric Wieling wrote: Try Hangup(123) where 123 is whatever hangup cause you want to send back to the caller. The calliing Asterisk server will get the valuse back in HANGUPCAUSE variable. Hello, I have tried sending Hangup(321) on Asterisk server B to Asterisk A but when I read HangupCause on Asterisk A it always is '21'. Good idea, but it does not seem to work. Kind regards, Jonas. -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
321 is not a valid Asterisk hangup cause. Valid hangupcauses are 1-127 (Q.831 cause codes) See https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, September 02, 2014 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk On 02-09-14 14:22, Eric Wieling wrote: Try Hangup(123) where 123 is whatever hangup cause you want to send back to the caller. The calliing Asterisk server will get the valuse back in HANGUPCAUSE variable. Hello, I have tried sending Hangup(321) on Asterisk server B to Asterisk A but when I read HangupCause on Asterisk A it always is '21'. Good idea, but it does not seem to work. Kind regards, Jonas. -- _ -- 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] PJSIP issues with handling incoming calls
Nick Awesome wrote: register = 73432260005:pass@10001 register = 73432260050:pass@10002 [10001] type=peer host=80.75.132.66 context=dialmap [10002] type=peer host=80.75.132.66 context=dialmap Can you provide a sip debug of calls to both of these? I'm confused how that... works... -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] Setup Own IP PBX Server
On 01-09-14 12:31, Chandran Manikandan wrote: [snip] I have installed Freepbx server and tried to configure sip extension. It's working fine. A better place for FreePBX related questions and to get help is: http://community.freepbx.org/ Or hire their professional FreePBX support: http://www.freepbx.org/support-and-professional-services If you want to learn more about Asterisk in general then a good start is to first read Asterisk: The Definitive Guide, 4th Edition and go through the wiki at http://wiki.asterisk.org. HTH, Patrick -- _ -- 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] POSSA CID Superfecta to query a CardDAV server ?
On 02/09/14 09:12, Lukasz Sokol wrote: On 01/09/14 12:05, Marie Fischer wrote: Well, you made me curious - wrote up a little perl script to do a filtered report by phone number. It takes 2-3 seconds to get a response from OSX server (Mavericks). Which sure is shorter then doing a full sync, but still longish. Would be interesting to know how long other servers take. I can try to time my CardDAV query program (keeping in mind that it runs on the same physical segment and subnet as the server) (sorry I don't know Perl ;) I'm a Free Pascal user ) Will come back to that. OK, alpha results of timing, show actually sending a query and receiving a response takes ~500 - 800 ms. Total response time for a query 1.3~2.2s max, if authentication required before query (but authentication request can be sent by means of some cron sending OPTIONS cookies every so often, to avoid session timeout; so counting that out, 500-800ms is what it takes for the server to answer the query). Now, for CID you would want this to run in your dial plan after the call comes in and before you Dial() your local extension. One ringtone is 5 seconds (1 sec tone, 4 sec silence), so it's actually not too bad (remember those analog caller ID boxes which got the caller ID between first and second ringtone?). Maybe you'd need to send Progress() or Ringing() back to the calling party. Hmm ok. Let me time it and see the timings. Keeping in mind this is going to be running on FreePBX 2.11.0.38, where do I start ;) if I wanted to marry this to CID Superfecta ? Or should I rather look for how to write my own FreePBX module ? You've mentioned 'LWP::UserAgent and HTTP::Request and Text::vCard::Addressbook' which I am going to research a bit. The fun part would be to make the response from CardDAV into an URL on which the operator shall click to bring the addressbook details on screen (or just call firefox with url given as param that would do already ;) ) Lukasz -- _ -- 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] PJSIP issues with handling incoming calls
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] PJSIP issues with handling incoming calls
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just
look at the cli output *NoOp( 49${gotoadr:-11} )
Am 02.09.2014 um 17:04 schrieb Rainer Piper:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP issues with handling incoming calls
upps and delete the 49 in *Goto(49${gotoadr:-11},1) and *NoOp(
49${gotoadr:-11} )
*just look at the cli output*
Am 02.09.2014 um 17:25 schrieb Rainer Piper:
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just
look at the cli output *NoOp( 49${gotoadr:-11} )
Am 02.09.2014 um 17:04 schrieb Rainer Piper:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- 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] PJSIP issues with handling incoming calls
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de wrote:
I use in pjsip.conf
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=sipgatefilter ; goto the filter in extensions.conf
retry_interval=60
forbidden_retry_interval=600
expiration=3600
extensions.conf ; i'm cutting the dialed number out of the invite Header and
goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,Goto(49${gotoadr:-11},1)
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten = 4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten = 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused how
that... works...
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- 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
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] PJSIP issues with handling incoming calls
contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi
script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de
mailto:rainer.pi...@soho-piper.de wrote:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls
so it seems Asterisk Versions does not support video I guess On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati khalidtou...@gmail.com wrote: Any article that goes through this (seems to be tedious) task to add video support and patents? On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp jc...@digium.com wrote: Khalid Touati wrote: Hi Guys, Kia ora, Do you know of any asterisk community version that does video codec trans-coding or in other words supports video? I have 1.8.8.1 and I see h263.c format files but can't see that codec in make menuselect. it might be just a license issue (if h263 has to have license), but not sure if community versions offer video calls at all. Video transcoding is both usually patent encumbered as well as computationally expensive. Asterisk supports passing through the video untouched, but that's about it. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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 -- Khalid Touati -- Khalid Touati -- _ -- 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] PJSIP issues with handling incoming calls
Am 02.09.2014 um 20:11 schrieb Rainer Piper: username ? -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test) -- _ -- 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] PJSIP issues with handling incoming calls
contact_user in pjsip.conf has to point to the filter or to an agi in
the extentions.conf
like:
pjsip.conf
contact_user=*blablabla
extensions.conf
**exten = blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
*
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi
script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de
mailto:rainer.pi...@soho-piper.de wrote:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- 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
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- 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] Does Asterisk 1.8. Supports Video Calls
On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
As long as you are NOT transcoding video should work in Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
On 02-09-14 21:15, Eric Wieling wrote: As long as you are NOT transcoding video should work in Asterisk. Both apps were configured with identical (codec) settings so I don't see how it would require transcoding. If you did get it to work I would appreciate it if you could tell me which clients you used, the Asterisk version, the OS and the relevant Asterisk config. Thanks, Patrick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
Le 02/09/2014 20:18, Khalid Touati a écrit : so it seems Asterisk Versions does not support video I guess Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 with GrandStream phones (H263, H263+ and H264). Works perfectly On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati khalidtou...@gmail.com mailto:khalidtou...@gmail.com wrote: Any article that goes through this (seems to be tedious) task to add video support and patents? On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: Khalid Touati wrote: Hi Guys, Kia ora, Do you know of any asterisk community version that does video codec trans-coding or in other words supports video? I have 1.8.8.1 and I see h263.c format files but can't see that codec in make menuselect. it might be just a license issue (if h263 has to have license), but not sure if community versions offer video calls at all. Video transcoding is both usually patent encumbered as well as computationally expensive. Asterisk supports passing through the video untouched, but that's about it. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com http://www.digium.com www.asterisk.org http://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 http://lists.digium.com/mailman/listinfo/asterisk-users -- Khalid Touati -- Khalid Touati -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
On 09/02/2014 03:14 PM, Administrator TOOTAI wrote: Le 02/09/2014 20:18, Khalid Touati a écrit : so it seems Asterisk Versions does not support video I guess Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 with GrandStream phones (H263, H263+ and H264). Works perfectly I can second that with GS phones, asterisk 1.4 and 1.8. j -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
A co-worker was doing video, I dislike video. The phones were Polycom VVX, The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and H.263p are the only two video codecs enabled. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 7:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 21:15, Eric Wieling wrote: As long as you are NOT transcoding video should work in Asterisk. Both apps were configured with identical (codec) settings so I don't see how it would require transcoding. If you did get it to work I would appreciate it if you could tell me which clients you used, the Asterisk version, the OS and the relevant Asterisk config. Thanks, Patrick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
You might want to check if videosupport=yes in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Tuesday, September 02, 2014 1:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls A co-worker was doing video, I dislike video. The phones were Polycom VVX, The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and H.263p are the only two video codecs enabled. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 7:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 21:15, Eric Wieling wrote: As long as you are NOT transcoding video should work in Asterisk. Both apps were configured with identical (codec) settings so I don't see how it would require transcoding. If you did get it to work I would appreciate it if you could tell me which clients you used, the Asterisk version, the OS and the relevant Asterisk config. Thanks, Patrick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- 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 -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
Don't forget videosupport=yes in sip.conf. j On 09/02/2014 03:52 PM, Eric Wieling wrote: A co-worker was doing video, I dislike video. The phones were Polycom VVX, The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and H.263p are the only two video codecs enabled. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 7:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 21:15, Eric Wieling wrote: As long as you are NOT transcoding video should work in Asterisk. Both apps were configured with identical (codec) settings so I don't see how it would require transcoding. If you did get it to work I would appreciate it if you could tell me which clients you used, the Asterisk version, the OS and the relevant Asterisk config. Thanks, Patrick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either H264 or VP8. HTH, Patrick -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
On 02-09-14 22:52, Eric Wieling wrote: A co-worker was doing video, I dislike video. The phones were Polycom VVX, The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and H.263p are the only two video codecs enabled. Thanks Eric. The obvious difference is that your co-worker was using H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present in my sip.conf so it might be the codec. Time for more tinkering. Thanks, Patrick -- _ -- 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] Does Asterisk 1.8. Supports Video Calls
core show codecs does not show VP8 on my Asterisk 11. I don't recall why we are not using H.264. The novelty wore off long ago and few of our staff use video calling anymore. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 9:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 22:52, Eric Wieling wrote: A co-worker was doing video, I dislike video. The phones were Polycom VVX, The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and H.263p are the only two video codecs enabled. Thanks Eric. The obvious difference is that your co-worker was using H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present in my sip.conf so it might be the codec. Time for more tinkering. Thanks, Patrick -- _ -- 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] Setup Own IP PBX Server
Hi Patrick, Thanks for your help. Let me try your advise and come back to you if need further any assistance. On Tue, Sep 2, 2014 at 9:41 PM, Patrick Laimbock patr...@laimbock.com wrote: On 01-09-14 12:31, Chandran Manikandan wrote: [snip] I have installed Freepbx server and tried to configure sip extension. It's working fine. A better place for FreePBX related questions and to get help is: http://community.freepbx.org/ Or hire their professional FreePBX support: http://www.freepbx.org/support-and-professional-services If you want to learn more about Asterisk in general then a good start is to first read Asterisk: The Definitive Guide, 4th Edition and go through the wiki at http://wiki.asterisk.org. HTH, Patrick -- _ -- 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 -- *Thanks,* *Manikandan.C* *System Administrator* -- _ -- 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] PJSIP issues with handling incoming calls
Ok, thanks for an answer. That solution works.
On 02 Sep 2014, at 22:36, Rainer Piper rainer.pi...@soho-piper.de wrote:
contact_user in pjsip.conf has to point to the filter or to an agi in the
extentions.conf
like:
pjsip.conf
contact_user=blablabla
extensions.conf
exten = blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)}
***)
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de wrote:
I use in pjsip.conf
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=sipgatefilter ; goto the filter in extensions.conf
retry_interval=60
forbidden_retry_interval=600
expiration=3600
extensions.conf ; i'm cutting the dialed number out of the invite Header
and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,Goto(49${gotoadr:-11},1)
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused how
that... works...
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
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--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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_
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