[asterisk-users] Regrading Speech Recognition.
Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks Akhilesh -- _ -- 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] Regrading Speech Recognition.
Things that look simple r quite complex to build :-) Indian Accent ASR on proper names is herculean task. No speech recognition known to mankind as of date can handle so many dialects being spoken in India, so in short what you want is nice to have, but nearly impossible to develop. Better try with short vocab on generic words (sales, support, etc.) Mitul On Jul 5, 2012 12:23 PM, akhilesh chand omakhileshch...@gmail.com wrote: Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks Akhilesh -- _ -- 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] Regrading Speech Recognition.
ok,how can i develop with short vocab like sales,support,etc. I have read many article but I'm not able to pick the right point, how can i develop or configure speech reorganization with asterisk. Is there any article or link please share and guide me. Regards Akhilesh On Thu, Jul 5, 2012 at 12:29 PM, Mitul Limbani mi...@enterux.in wrote: Things that look simple r quite complex to build :-) Indian Accent ASR on proper names is herculean task. No speech recognition known to mankind as of date can handle so many dialects being spoken in India, so in short what you want is nice to have, but nearly impossible to develop. Better try with short vocab on generic words (sales, support, etc.) Mitul On Jul 5, 2012 12:23 PM, akhilesh chand omakhileshch...@gmail.com wrote: Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks Akhilesh -- _ -- 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
[asterisk-users] Elastix 2.3.0.1
Greetings, I know this is not a Elastix mailing list, but could anybody please tell where I can download Elastix 2.3.0.1 (the latest version) ? There is only version 2.3.0 (April 2012) on Elastix website, not the 2.3.0.1 (May 2012), but the changelog information are there. Thanks in advance. Best regards, Anam. -- _ -- 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] Timer1 RFC and SIP.CONF
4 jul 2012 kl. 13:32 skrev Elliot Murdock: Hello, I am trying to get clarity with the sip.conf timer configuration. The current configuration states: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 However, according to RFC 3261: (EXCERPT 17.1.1.1) T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 adjusts their values. The request is not retransmitted over reliable transports. After receiving a 1xx response, any retransmissions cease altogether, and the client waits for further responses. The server transaction can send additional 1xx responses, which are not transmitted reliably by the server transaction. Eventually, the server transaction decides to send a final response. (EXCERPT 13.2.2.4 2xx Responses) The UAC core considers the INVITE transaction completed 64*T1 seconds after the reception of the first 2xx response. According to the RFC, the 64*t1 timeout is not for provisional responses, but for final responses. This seems to be in contradiction to what is stated in the sip.conf file. Unless you have PRACK support, which Asterisk not yet has, there's no timeout in the SIP protocol for provisional responses more than the need to update your provisional response at least every minute not to hit a 3 minute timeout in the SIP transaction state in a proxy. Now, the timerb is used to get ANY response from a server out there, including provisional responses. If we don't get ANY response within TIMERB, the SIP transaction dies and in a UA with a transaction layer we generate a local 408 timeout. In Asterisk, we congest the call. So the 64*T1 is for any response, including final response. It's there to decide whether or not you have intelligent SIP life forms handling your SIP request in the network universe. I hope this clears up your confusion. Regards, /Olle -- _ -- 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 set debug on always showing error
dear please Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side. --- Transmitting (NAT) to 122.163.193.94:1893 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893 From: 2002 sip:2002@122.160.154.189;tag=5a1cc54c To: 2002 sip:2002@122.160.154.189;tag=as64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS -- _ -- 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] sip set debug on always showing error
Hi, *CSeq: 245 OPTIONS * * * This is just SIP keep-alive. It has nothing to do with any Call-media degradation. If you are not getting clear voice check the codecs, network latency/delay/loss/jitter parameters. BR Sammy On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava alok...@gmail.com wrote: dear please Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side. --- Transmitting (NAT) to 122.163.193.94:1893 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893 From: 2002 sip:2002@122.160.154.189;tag=5a1cc54c To: 2002 sip:2002@122.160.154.189;tag=as64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS -- _ -- 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] sip set debug on always showing error
previously i was using for codec allow=all after that i changed disallow=all allow=silk24 and i also change softph x-lite from jitsi(because of codec) now voice was coming fine from both side. But when i came to home from office not getting voice from both side. Threr is Airtel Broadband at my place. On Thu, Jul 5, 2012 at 3:36 PM, SamyGo govoi...@gmail.com wrote: Hi, *CSeq: 245 OPTIONS * * * This is just SIP keep-alive. It has nothing to do with any Call-media degradation. If you are not getting clear voice check the codecs, network latency/delay/loss/jitter parameters. BR Sammy On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava alok...@gmail.com wrote: dear please Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side. --- Transmitting (NAT) to 122.163.193.94:1893 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893 From: 2002 sip:2002@122.160.154.189;tag=5a1cc54c To: 2002 sip:2002@122.160.154.189;tag=as64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS -- _ -- 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] basic sip quesiton
Put disallow=all below all of the allow= -- _ -- 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] OT - Integration with building intercom systems
Hi, Now and then, I'm facing environments in which it could be helpful to integrate building intercom systems with Asterisk. Those intercom systems are made of : - a main panel, showing company names and equiped with a speaker, a microphone and an optional video cam - a doorstrike - several intercom phone with an optional monitor and with a button triggering door's opening, - maybe other hidden components I'm not aware of. I've noticed that these intercom phone are connected to the main panel through a 4 or 5-wires cable. More precisely, for the video case I have in mind, it's a 4-wires cable: 2 thick wires (for energy ?) and 2 thin wires (for voice, video and command). Is there a standardized protocol available to run voice, video and command on 2-wires and most probably used by intercom systems ? Would you say it's possible to install several intercom phones on the same line, both ringing at the same time (but only one of them being able to answer) ? If positive, is it possible to connect one intercom line to asterisk and let an asterisk user talk or watch visitor before opening the door (through a DTMF sequence, for instance) ? Regards -- _ -- 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] OT - Integration with building intercom systems
Le 05/07/2012 15:21, Olivier a écrit : [...] Would you say it's possible to install several intercom phones on the same line, both ringing at the same time (but only one of them being able to answer) ? If positive, is it possible to connect one intercom line to asterisk and let an asterisk user talk or watch visitor before opening the door (through a DTMF sequence, for instance) ? Sure you can. Any intercom can be connected using a IP gw connected to asterisk. You then install an IP camera and it's done :-) Opening the door is done by sending the code using asterisk sendDTMF command. That's what we do when using existing intercom. For new installations we use Mobotix products (IP doorphone in SIP). -- Daniel -- _ -- 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] OT - Multi Function Printer with one touch scanning/emailing
Hi, I'm curious about the availability of Multi Function Printers with the following feature : - user feeds paper sheets in - user dials a phone number (0123456, for instance) then a hits single button - the result is that the paper sheets are scanned into a file which is emailed to a given address such as 0123...@myfaxgateway.com (where myfaxgateway.com is a fixed and configured address). Is this a common feature ? Last time I checked, MFP's alphanumeric diaplan was either oriented to digits or letters typing, and of course, scanning feature implied letters typing mode. Regards -- _ -- 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] Elastix 2.3.0.1
El 05/07/12 02:19, Satria Anamarta escribió: Greetings, I know this is not a Elastix mailing list, but could anybody please tell where I can download Elastix 2.3.0.1 (the latest version) ? There is only version 2.3.0 (April 2012) on Elastix website, not the 2.3.0.1 (May 2012), but the changelog information are there. You can always try installing 2.3.0 then updating everything with yum update. -- _ -- 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] OT - Multi Function Printer with one touch scanning/emailing
T.37 http://en.wikipedia.org/wiki/T.37_(ITU-T_recommendation) There were some scanners manufactured with this in mind, however I cant remember who made them. On Thu, Jul 5, 2012 at 10:15 AM, Olivier oza_4...@yahoo.fr wrote: Hi, I'm curious about the availability of Multi Function Printers with the following feature : - user feeds paper sheets in - user dials a phone number (0123456, for instance) then a hits single button - the result is that the paper sheets are scanned into a file which is emailed to a given address such as 0123...@myfaxgateway.com (where myfaxgateway.com is a fixed and configured address). Is this a common feature ? Last time I checked, MFP's alphanumeric diaplan was either oriented to digits or letters typing, and of course, scanning feature implied letters typing mode. Regards -- _ -- 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] OT - Multi Function Printer with one touch scanning/emailing
I searched a bit more, http://www.muratec.com/catalog/F320_config.html#email The above model supports t.37 but no sure if you can have it function such that any number entered will actually be send to a gateway. On Thu, Jul 5, 2012 at 10:20 AM, C F shma...@gmail.com wrote: T.37 http://en.wikipedia.org/wiki/T.37_(ITU-T_recommendation) There were some scanners manufactured with this in mind, however I cant remember who made them. On Thu, Jul 5, 2012 at 10:15 AM, Olivier oza_4...@yahoo.fr wrote: Hi, I'm curious about the availability of Multi Function Printers with the following feature : - user feeds paper sheets in - user dials a phone number (0123456, for instance) then a hits single button - the result is that the paper sheets are scanned into a file which is emailed to a given address such as 0123...@myfaxgateway.com (where myfaxgateway.com is a fixed and configured address). Is this a common feature ? Last time I checked, MFP's alphanumeric diaplan was either oriented to digits or letters typing, and of course, scanning feature implied letters typing mode. Regards -- _ -- 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] OT - Integration with building intercom systems
Il 05/07/2012 15.21, Olivier ha scritto: Now and then, I'm facing environments in which it could be helpful to integrate building intercom systems with Asterisk. Is there a standardized protocol available to run voice, video and command on 2-wires and most probably used by intercom systems ? I think this is country specific, I know a quite large number of de facto standards but almost no real standard except the German FTZ 123 D 12. Most door phone analog system here (Italy) consists of 3/4+1 (bus + individual ring signal) wires and some more recent using proprietary digital protocols on a 2 wires bus. Sure I seen something similar in France. Interfacing any old individual analog door phone to asterisk isn't so difficult using a door phone to fxo/fxo adapter like used to do with a legacy pbx. The matter becomes more difficult approaching a building install as there are no devices to handle properly that. I think the snom PA-1 may be a good candidate to play with because of a versatile I/O that could be interfaced to a custom door-phone bridge to IP. -- _ -- 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] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
Hi Shitian, here's my sip.conf, but unfortunately I cannot make some other tests with Asterisk 1.8 since the PBX is in production now with Asterisk 1.4.26.2 which seems to work very fine. Thank you G NOTE: tried to change nat and canreinvite parameters but with no success. [general] disallow = all allow = alaw allow = ulaw allow = g726 allow = g723.1 allow = gsm notifyringing = yes limitonpeer = yes notifyhold = yes monitor-format = wav musicclass = default callerid = unknown callcounter = yes allowguest = no context = inbound busylimit = 1 srvlookup = no port = 5060 transport = udp bindaddr = 0.0.0.0 notifybusy = yes register = 123456789:pas...@psip1.mclink.it:5060/123456789 ; [123456789] ; Options from provider (provider.sip-mclink) host = psip1.mclink.it nat = yes canreinvite = yes type = peer context = outbound qualify = yes port = 5060 fromdomain = psip1.mclink.it insecure = very language = it fromuser = 123456789 username = 123456789 secret = passwd On 07/02/2012 12:32 AM, Shitian Long wrote: if you check out your sip.conf. On Jun 29, 2012, at 5:54 PM, gincantalupo wrote: Hi all, after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150. The strange thing is that the rport inside SIP packets (sip set debug) coming back from my provider is set to 55150.seen on both Asterisk 1.4 and 1.8 Does anybody have any idea? Thank you. Giorgio -- _ -- 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] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
Il 05/07/2012 16.43, gincantalupo ha scritto: here's my sip.conf, but unfortunately I cannot make some other tests with Asterisk 1.8 since the PBX is in production now with Asterisk 1.4.26.2 which seems to work very fine. I'm using the same provider on many sites without special issues. My sip.conf follows, tested time ago on 1.4, ported with minor changes to 1.6.2 (now in production) then ported to 1.8 without changes (lab test only). [general] context=public-direct-dialin allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes useragent=TeeBX alwaysauthreject=yes videosupport=no notifybusy=yes counteronpeer=yes notifyhold=no pedantic=yes callcounter=yes defaultexpiry=120 minexpiry=60 maxexpiry=3600 localnet=172.31.255.0/24 localnet=172.31.254.0/24 ; MCLink register = username:p...@psip1.mclink.it/username [mclink-06x] type=peer defaultuser=username secret=pass fromuser=username host=psip1.mclink.it context=mclink-06x-incoming fromdomain=psip1.mclink.it language=it-it nat=yes qualify=2000 directmedia=no insecure=port,invite dtmfmode=rfc2833 disallow=all allow=alaw allow=gsm call-limit=5 -- _ -- 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] touch command not behaving for future calls in asterisk 1.4.41
Hi All, It's small issue but making a big problem for my application. I have CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41 because Flite work in this version. problem is that when I make changes on .call file to make it future call file with *touch *command then it not changed. [root@server tmp]# touch -t 201207052137 1341509545.39.call [root@server tmp]# ll -rw-r--r-- 1 root root 52 Jul 5 2012 1341509545.39.call .call file's time is missed with year only that's asterisk make call after move to outgoing folder. please give your suggestion. If I am wrong then correct me ... -- Thanks and regards Virendra Bhati +91-9718300881 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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] Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, 10.5.2-digiumphones Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones resolve the following two issues: * If Asterisk sends a re-invite and an endpoint responds to the re-invite with a provisional response but never sends a final response, then the SIP dialog structure is never freed and the RTP ports for the call are never released. If an attacker has the ability to place a call, they could create a denial of service by using all available RTP ports. * If a single voicemail account is manipulated by two parties simultaneously, a condition can occur where memory is freed twice causing a crash. These issues and their resolution are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2012-010 and AST-2012-011, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert4 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2-digiumphones The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2012-010.pdf * http://downloads.asterisk.org/pub/security/AST-2012-011.pdf Thank you for your continued support of Asterisk! -- _ -- 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] Outbound Asterisk calls default directmedia specifications
On 07/04/2012 01:47 PM, sathiish kumar wrote: Thanks for the response.. I did change it in the [general] settings.My setup is something like I have a remote conference (not meetme) which will send reinvite to redirect the RTP flow to a different server to load balance.There are three clients who join in the conference and i can listen to two other clients speak from the third client but when i record the conversation my recording of one of the clients ends before the stipulated hangup time. I am guessing this is because one of the clients doesn't understand what to do with a reinvite.. Any suggestions.In the SIP.conf i have changed the directmedia option to no and also enabled the ignoresdpversion option. The 'directmedia' option *only* controls whether Asterisk will attempt to drop itself out of the media path between two SIP endpoints. It has no effect on whether or not Asterisk will respond appropriately to a re-INVITE received *from* a SIP endpoint (to which Asterisk should always respond properly, unless the re-INVITE is malformed in some way or is unacceptable). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] sip and extensions
I am new. Here is the code that I am playing with on CentOS 6.x When I dial the number that corresponds w/ my SIP account I get a recording: reached a non-working number I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf I don't remember using a file that long to get a basic call set up. The format was provided to me by voipvoip.com (the SIP provider). Does anyone have any comments please? I just want a very simple config to get my machine to recognize a call to the SIP provider. Here is results of sip show registry: Hostdnsmgr Username Refresh StateReg.Time sip3.voipvoip.com:5060 N 5552530146 285 Registered Thu, 05 Jul 2012 21:39:56 1 SIP registrations. Here is sip and extensions.conf sip.conf [general] register = 5552530146:funnytiger...@sip3.voipvoip.com ; [sip3.voipvoip.com] [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no extensions.conf [general] ; ; [incoming] ;first creating extensions for your local users exten= s,1,Dial(SIP/1703717) exten= s,2,Hangup() -- _ -- 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] sip and extensions
- Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register = 5552530146:funnytiger...@sip3.voipvoip.com [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no *PLEASE* if that is your real username/password with your VoIP provider change it immediately. Just FYI, you've broadcast it to (tens or hundreds of) thousands of list readers. I have to believe some are of the nefarious type that would love to use your account for free calling at your expense. :/ --Tim -- _ -- 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] sip and extensions
Hi, I changed these codes to not coincide with actual account info. Thanks On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register = 5552530146:funnytiger...@sip3.voipvoip.com [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no *PLEASE* if that is your real username/password with your VoIP provider change it immediately. Just FYI, you've broadcast it to (tens or hundreds of) thousands of list readers. I have to believe some are of the nefarious type that would love to use your account for free calling at your expense. :/ --Tim -- _ -- 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] AMR - Segmentation Fault
On Wed, 2012-07-04 at 10:15 +0530, Chandrakant Solanki wrote: So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch also works in 1.8.13.0?? On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet aster...@a-domani.nl wrote: On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote: Hi All, OS : Cent OS 5 64Bit Asterisk : 1.8.0-rc2 AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/ When I tried to call or start asterisk, I found Segmentation Fault. Without trying to be pedantic, but 1.8.0-rc2 Ever considered upgrading? To 1.8.13.0 or so.. Hm, i see. Looks like somebody is seriously hibernating: 1.8.0-rc2_asterisk_amr_patch.diff 2010-10-15 Almost two years old! If it is only the codec itself, you might try: http://ftp5.gwdg.de/pub/linux/packman/suse/12.1/Multimedia/src/amrnb-10.0.0.0-1.1.src.rpm or http://ftp5.gwdg.de/pub/linux/packman/suse/12.1/Essentials/src/amrwb-10.0.0.0-1.1.src.rpm As these are source packages, you might be able to turn then into deb's or so 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
[asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I have to write the configuration in my hand and it will not be autogeneration, correct? In this case, the Phone will not have any of the features that I am going to add it in the GUI because these features will be in the default context which is not included (unless I add it manually) in the context that I will set it. Also, if I set the context and I write manually the configuration for this context, I do not think that I will have CDR (because to have CDR, I have to use some configuration to log in the database and becoming able to see it in the CDR). Again, if I used the default context, then it is good that all the stations to have the same context and same previlages .. so it is not a practical way. So, what is the solution for this? As I see the only benifit of the Freepbx (the GUI), is to generate the configuration that I can use it when I am writing the manual configuration (by including it and so on). In this case, I am afraid that things will become maybe more complex :) !! Any advise for this? Regards Bilal -- _ -- 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] FreePBX: How to hangup if the caller did not press # after the voicemail message
Dears; In FreePBX, when I select voicemail for the extension, and if the caller sent for the voicemail, and he leaved (or did not leave) a voice message, and did not press #, so the channel will stay open and this is not good specially if the call was coming from outside via the analoge lines (because the caller might hangup and the dahdi does not detect the hangup, so the channel will stay openned). How to let the voicemail hangup automatically after waiting for certain seconds (for example after 30 or 40 second), then to hangup or jump for the next line to run it? What is the parameter or the setting field in the freepbx that can resolve this (the voice mail message to be maximum for 30 or 40 second, after that to hangup even without pressing #). From where? Regards Bilal -- _ -- 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] FreePBX: How to hangup if the caller did not press # after the voicemail message
Asterisk, and by extension FreePBX, automatically end the voicemail recording when the caller hangs up. You have some OTHER issue. Perhaps Asterisk is not detecting the hangup? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, July 05, 2012 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message Dears; In FreePBX, when I select voicemail for the extension, and if the caller sent for the voicemail, and he leaved (or did not leave) a voice message, and did not press #, so the channel will stay open and this is not good specially if the call was coming from outside via the analoge lines (because the caller might hangup and the dahdi does not detect the hangup, so the channel will stay openned). How to let the voicemail hangup automatically after waiting for certain seconds (for example after 30 or 40 second), then to hangup or jump for the next line to run it? What is the parameter or the setting field in the freepbx that can resolve this (the voice mail message to be maximum for 30 or 40 second, after that to hangup even without pressing #). From where? Regards Bilal -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
Hey, If you want to have all the dialplan features for your extensions and still need to implement some outbound calling restrictions then you need to look for some modules in freePBX. i've used that module exactly for this purpose and it works..can't remember its name. Just google it or lookup the latest modules available. Regards, Sammy On Fri, Jul 6, 2012 at 3:20 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I have to write the configuration in my hand and it will not be autogeneration, correct? In this case, the Phone will not have any of the features that I am going to add it in the GUI because these features will be in the default context which is not included (unless I add it manually) in the context that I will set it. Also, if I set the context and I write manually the configuration for this context, I do not think that I will have CDR (because to have CDR, I have to use some configuration to log in the database and becoming able to see it in the CDR). Again, if I used the default context, then it is good that all the stations to have the same context and same previlages .. so it is not a practical way. So, what is the solution for this? As I see the only benifit of the Freepbx (the GUI), is to generate the configuration that I can use it when I am writing the manual configuration (by including it and so on). In this case, I am afraid that things will become maybe more complex :) !! Any advise for this? Regards Bilal -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
The module is custom contexts - its a third party option in the module admin But you can write contexts in the extensions_custom.conf if you want to I wouldn't use freepbx to generate your code - its quite complex code for a roll your own system, but very useful if you learn its gui and options Also you can limit outbound routes to certain extension ranges which can avoid the need for contexts but its up to you Cheers Duncan On 6/07/2012, at 4:20 PM, SamyGo wrote: Hey, If you want to have all the dialplan features for your extensions and still need to implement some outbound calling restrictions then you need to look for some modules in freePBX. i've used that module exactly for this purpose and it works..can't remember its name. Just google it or lookup the latest modules available. Regards, Sammy On Fri, Jul 6, 2012 at 3:20 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I have to write the configuration in my hand and it will not be autogeneration, correct? In this case, the Phone will not have any of the features that I am going to add it in the GUI because these features will be in the default context which is not included (unless I add it manually) in the context that I will set it. Also, if I set the context and I write manually the configuration for this context, I do not think that I will have CDR (because to have CDR, I have to use some configuration to log in the database and becoming able to see it in the CDR). Again, if I used the default context, then it is good that all the stations to have the same context and same previlages .. so it is not a practical way. So, what is the solution for this? As I see the only benifit of the Freepbx (the GUI), is to generate the configuration that I can use it when I am writing the manual configuration (by including it and so on). In this case, I am afraid that things will become maybe more complex :) !! Any advise for this? Regards Bilal -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
On Thu, Jul 5, 2012 at 5:20 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; You can get modules to do what you're looking for, but if you really want to make a custom context but still have all the available features of the default context, you can add the following at the end of your custom context: include = from-internal Be sure to do all of this in extensions_custom.conf, that way it doesn't get overwritten whenever you issue a reload in the GUI. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
umm Warren, yes including from-internal is the way of getting all the features,,,but in my experience the calls going out using the dialplan script we manually enter in our custome context don't get inserted into the FreePBX CDR and recording stuff !! On Fri, Jul 6, 2012 at 10:01 AM, Warren Selby wcse...@selbytech.com wrote: On Thu, Jul 5, 2012 at 5:20 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; You can get modules to do what you're looking for, but if you really want to make a custom context but still have all the available features of the default context, you can add the following at the end of your custom context: include = from-internal Be sure to do all of this in extensions_custom.conf, that way it doesn't get overwritten whenever you issue a reload in the GUI. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
On Fri, Jul 6, 2012 at 12:11 AM, SamyGo govoi...@gmail.com wrote:
umm Warren, yes including from-internal is the way of getting all the
features,,,but in my experience the calls going out using the dialplan
script we manually enter in our custome context don't get inserted into the
FreePBX CDR and recording stuff !!
Okay, if you're writing custom dialplan to control outbound calling, but
you want to utilize the FreePBX standard features, without using custom
modules, you can do something like the following, adjusting for your
specific situations of course:
[custom-local-only]
; local NANPA calling for area code 281
exten = _281NXX,1,Verbose(Outbound call from local-only context)
same = n,Goto(${EXTEN},from-internal,1)
; extension-to-extension (internal) calling, assuming 2XXX internal
extension plan
exten = _2XXX,1,Verbose(Internal extension-to-extension call)
same = n,Goto(${EXTEN},from-internal,1)
[custom-long-distance]
; long distance NANPA calling, dial a 1 to dial anything outside of a local
number
exten = _1NXXNXX,1,Verbose(Outbound call from local and long-distance
context)
same = n,Goto(${EXTEN},from-internal,1)
; allow local calls also, without having to dial a 1
include = custom-local-only
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
--
_
-- 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] touch command not behaving for future calls in asterisk 1.4.41
Hi, Did you get anything working on it !! See the permission for the user running asterisk process and see if that user can touch files like that. Regards, Sammy On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.com wrote: Hi All, It's small issue but making a big problem for my application. I have CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41 because Flite work in this version. problem is that when I make changes on .call file to make it future call file with *touch *command then it not changed. [root@server tmp]# touch -t 201207052137 1341509545.39.call [root@server tmp]# ll -rw-r--r-- 1 root root 52 Jul 5 2012 1341509545.39.call .call file's time is missed with year only that's asterisk make call after move to outgoing folder. please give your suggestion. If I am wrong then correct me ... -- Thanks and regards Virendra Bhati +91-9718300881 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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] touch command not behaving for future calls in asterisk 1.4.41
Thanks Gohar, I found the issue was copy file to outbound folder not moving. that's why after making future time asterisk start reading file. On Fri, Jul 6, 2012 at 11:16 AM, SamyGo govoi...@gmail.com wrote: Hi, Did you get anything working on it !! See the permission for the user running asterisk process and see if that user can touch files like that. Regards, Sammy On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.comwrote: Hi All, It's small issue but making a big problem for my application. I have CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41 because Flite work in this version. problem is that when I make changes on .call file to make it future call file with *touch *command then it not changed. [root@server tmp]# touch -t 201207052137 1341509545.39.call [root@server tmp]# ll -rw-r--r-- 1 root root 52 Jul 5 2012 1341509545.39.call .call file's time is missed with year only that's asterisk make call after move to outgoing folder. please give your suggestion. If I am wrong then correct me ... -- Thanks and regards Virendra Bhati +91-9718300881 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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
