[asterisk-users] google voice + asterisk: calls made to GV# processed but weird
Dear all, First off, I am very new to asterisk so forgive me if any of my comments or questions seem trivial. Thanks to [this post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/) and [this post](http://www.davidvossel.com/?p=28), I have GV set up on asterisk through jabber.conf and gtalk.conf. I can successfully dial out from asterisk. I'm trying to set up an auto-attendant on asterisk. I am doing a basic Hello world example. My config: jabber.conf: [general] debug=yes autoprune=no autoregister=yes [asterisk] type=client serverhost=talk.google.com username=mya...@gmail.com/gmail secret=MYPASSWORD port=5222 usetls=yes usesasl=yes statusmessage=Connected to Asterisk. ;required do not change timeout=100 gtalk.conf: [general] context=default allowguest=yes bindaddr=0.0.0.0 [guest] disallow=all allow=ulaw connection=asterisk extensions.conf: [general] [globals] [incoming] exten = s,1,Answer() exten = s,n,Playback(hello-world) exten = s,n,Hangup() [default] include = incoming Basically, when I'm logged into another gmail account and call the computer that's connected to asterisk, the Hello world example works. However, if I call the GV # from a phone, GV rings and end up at the GV voicemail. At first I thought it just skipped the pickup altogether. However, thanks to the help of p3nguin, pabelanger, and [TK]D-Fender on #asterisk, I found out that the call IS processed by asterisk; however, the user does not hear any of it and goes straight to the GV voicemail. I wanted to give the mailing list a try to see if other people have thoughts on this. Here is the debug: [Oct 24 21:18:23] VERBOSE[2393] config.c: == Parsing '/etc/asterisk/logger.conf': [Oct 24 21:18:23] DEBUG[2393] config.c: Parsing /etc/asterisk/logger.conf [Oct 24 21:18:23] VERBOSE[2393] config.c: == Found [Oct 24 21:18:23] VERBOSE[2393] logger.c: Asterisk Queue Logger restarted [Oct 24 21:18:28] VERBOSE[2405] res_jabber.c: JABBER: Keep alive packet [Oct 24 21:18:44] VERBOSE[2405] res_jabber.c: JABBER: asterisk INCOMING: presence from=cal...@gmail.com/androidfe2b05b6ebb0 to=myusern...@gmail.compriority24/prioritycaps:c node=http://www.android.com/gtalk/client/caps; ext=pmuc-v1 ver=1.1 xmlns:caps=http://jabber.org/protocol/caps/status/x xmlns=vcard-temp:x:updatephoto3c4fd5045a18d7417b2e4371bdce077ecd6c8355/photo/x/presence [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: JABBER: I am available ^_* 13 [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: JABBER: type is available [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: JABBER: Handling paktype PRESENCE [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: XML parsing successful [Oct 24 21:18:49] VERBOSE[2405] res_jabber.c: JABBER: asterisk INCOMING: iq from=+1caller10di...@voice.google.com/srvres-MTAuMjE4LjIwLjE0Mzo5ODEy to=myusern...@gmail.com/gmail02D370A8 id=jingle:10.218.20.143-28982014:1:C3955FF7 type=setses:session type=initiate id=sip183646...@10.218.118.3 initiator=+1caller10di...@voice.google.com/srvres-MTAuMjE4LjIwLjE0Mzo5ODEy xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq [Oct 24 21:18:49] DEBUG[2405] res_jabber.c: JABBER: Handling paktype IQ [Oct 24 21:18:49] DEBUG[2405] chan_gtalk.c: The client is guest for alloc [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] res_rtp_asterisk.c: Allocated port 11262 for RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: RTP instance '0x1b86bc8' is setup and ready to go [Oct 24 21:18:49] DEBUG[2405] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Setting payload 0 based on m type on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Setting payload 101 based on m type on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Incorporating payload 0 on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Incorporating payload 101 on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2399] devicestate.c: No provider found, checking channel drivers for Gtalk - +1CALLER10DIGIT [Oct 24 21:18:49] DEBUG[2399] devicestate.c: Changing state for Gtalk/+1CALLER10DIGIT - state 2 (In use) [Oct 24 21:18:49] DEBUG[2399] devicestate.c: device 'Gtalk/+1CALLER10DIGIT' state '2' [Oct 24 21:18:49] VERBOSE[4341] pbx.c: == Starting Gtalk/+1CALLER10DIGIT-12d0 at default,myusern...@gmail.com,1 failed so falling back to exten 's' [Oct 24 21:18:49] DEBUG[4341] pbx.c: Launching 'Answer' [Oct 24 21:18:49] VERBOSE[4341] pbx.c: -- Executing [...@default:1] Answer(Gtalk/+1CALLER10DIGIT-12d0, ) in new stack [Oct 24 21:18:49] DEBUG[2434] app_queue.c: Device
Re: [asterisk-users] E1 configuration
although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.comwrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com msn%3aflaviormira...@hotmail.com Skype: flaviormiranda -- From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com msn%3aflaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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] How to read core-en_US.xml
Hi. There is /doc/core-en_US.xml in asterisk 1.8 source tree. Is this file generated from documentation comments of apps/app_*.c files? And how this file can be used? How can I convert it to pdf/html in order to use it as applications documentation? -- _ -- 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] Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method
Am Donnerstag, den 21.10.2010, 16:27 +0200 schrieb Karsten Wemheuer: Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user C The call is terminated by asterisk without any warnings or error message in the CLI. Looking at Events on AMI, I can see in the first case an Event Newchannel with a Channel: AsyncGoto... followed by an Event Masquerade in prior to the Transfer. These events are missing in the second case. Is this a new bug or do I something wrong? Shall I open an issue on the tracker? for the archives: This behavior is still the same in 1.8.0. There is a workaround available (issue #18185) Karsten -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
I would probably do this through the AMI, it should spew out the info you require. Timestamp when they entered the queue and timestamp when they get answered. On 10/25/2010 05:01 AM, Bruce B wrote: Anything on this guys? I am sure someone had the need to record the HOLD time or maybe it is already being recorded somewhere? Any thoughts are appreciated. Thanks, Bruce On Wed, Oct 20, 2010 at 3:30 AM, Bruce B bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Hi Everyone, We are using Queuemetrics but it doesn't Record the Hold Time as it's never logged on the queue_log file. However, when an agent or an extension presses HOLD button on their phone, asterisk does create an event for Music On Hold which is logged in the /var/log/asterisk/full. I want to record the total hold time for an extension and save it with an epoch time stamp. What is the best approach to this? read and parse /var/log/asterisk/full in a cron job every few seconds? Have a presistent PHP-AGI connection to check for hold time events? As much detail as possible on above approaches or other ideas are most appreciated. Thanks p style=margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif; font-size: 10px; color: #33; ---DISCLAIMERbr / The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free./p -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Mon, Oct 25, 2010 at 12:50 AM, Anthony Messina amess...@messinet.com wrote:
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
; inbound ipKall trunk (to which Google Voice makes the connection)
[ipkall]
exten = ipKall-number,1,GotoIf($[${DB_EXISTS(gv/channel)} = 1]?gv)
same = n,Goto(default,s,1)
same = n(gv),Bridge(${DB_DELETE(gv/channel)})
same = n,AGI(gv/gv.agi,hangup)
same = n,Hangup()
; outbound Google Voice initiation
[gv-out]
exten = _X.,1,AGI(gv/gv.agi,call)
same = n,While($[${DB_EXISTS(gv/channel)} = 1])
same = n,Wait(0.3)
same = n,EndWhile()
same = n,Hangup()
And the AGI (written in Bash) is here:
http://messinet.com/trac/wiki/AsteriskGVGateway
http://messinet.com/trac/browser/gv/gv.agi
Does the AGI have to be used? In this example
http://www.davidvossel.com/?p=28 I see mention of a script, but not in
this one: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
I believe I missing the connection in how the whole process actually
works therefore making troubleshooting a little difficult. I was
hoping with the release of 1.6.0 there wouldn't be a lot of bandage
work to get it to play nicely with Google Voice.
--
_
-- 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] B410P - BRI NT 100 Ohm terminator
On Sat, Oct 23, 2010 at 02:07:27PM +0200, Olivier wrote: Hi, My set up is : Asterisk with B410P in NT mode -cat5 straight cable Another PBX in TE mode Is the 100 Ohm terminator you can find on B410P boards, necessary when connecting in NT mode to another PBX (set in TE mode) ? For a single connection? No. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] How to properly re-configure dahdi
On Sat, Oct 23, 2010 at 02:43:26PM +0200, Olivier wrote: Hi, How to properly re-configure dahdi, when for instance I want to change from TE to NT mode ? I'm planning the following operations : /etc/init.d/asterisk stop /etc/init.d/dahdi stop rmmod dahdi rm /etc/asterisk/dahdi-channels.conf rm /etc/dahdi/system.conf rm /etc/dahdi/modules nano /etc/dahdi/genconf_parameters dahdi_genconf -v modules dahdi_genconf -v system dahdi_genconf -v chandahdi Am I missing something ? I'm not sure that dahdi_genconf can tell those changes, as the driver does not report them. The modules file has not changed, anyway. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] CDR updating
Hi, I am using Asterisk 1.6.2.13 and have an issue but I'm not sure if it's a bug or not. I am using the cdr_adaptive_odbc logging module and writing my CDR records to a MS-SQL server. I need to log which end hangs the call up and have placed the relevant CDR(myfield)=caller/callee commands where they need to be. When I watch the call on the console I can see the CDR field being set properly but when I check the CDR record it is incorrect. It appears that when one end hangs up the CDR is being written immediately instead of waiting until the h exten. I have had a look in cdr.conf and set endbeforehexten=no, but this doesn't seem to make any difference. Does anyone have any ideas or is it a problem with the cdr_adaptive_odbc module? Thanks Lee -- _ -- 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] B410P - BRI NT 100 Ohm terminator
On 10/23/2010 07:07 AM, Olivier wrote: Hi, My set up is : Asterisk with B410P in NT mode -cat5 straight cable Another PBX in TE mode Is the 100 Ohm terminator you can find on B410P boards, necessary when connecting in NT mode to another PBX (set in TE mode) ? If you have it set up in a point-to-point mode, you do not need the termination. It is required when you are daisy chaining devices (like when using point-to-multipoint i.e. you are connecting several ISDN phones to one B410P in NT mode). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 configuration
Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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] Problem
On 10/23/2010 07:35 AM, ali raza wrote: Hello I am working on TDM2400p. I am having some problems like: when i connect my analog phone with the card there is no dial tone, but i can dial any extension... but after that i can't hear any voice from my receiver i have used different phone sets but still i cant communicate with other extension. Please help me out. There isn't enough information in this email to offer help. It sounds like there is one-way audio from the FXS port (assuming that when you said you're connecting your analog phone you mean you're connecting a handset to an FXS port). What is the output from dmesg when you load the driver? What version of the driver/asterisk are you using? Perhaps you could use dahdi_monitor to record the audio from the channel your phone is connected to in order to isolate the problem to either Asterisk or the drivers. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing muting enter the phone number
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH Sent: Saturday, October 23, 2010 7:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] a2billing muting enter the phone number How can I mute the message please enter the number you wish to call and press the # key in a2billing??? I tried use_dnid = YES but still I keep getting the message prompt... thanks the CLI is your friend here; when this prompt is playing, look at the CLI console and identify this file. It will be in /var/lib/asterisk/sounds. Replace it with beep.gsm. -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
Thanks for the feedback. I don't need the Queue times but rather putting ON HOLD times. If you press the HOLD button on your SIP phone, Asterisk records the event Music On HOLD Playing and that is recorded in /var/log/asterisk/full. I want to harvest the ON HOLD time per phone SET. Thanks On Mon, Oct 25, 2010 at 4:51 AM, Antonio Berrios anto...@sheffieldcitytaxis.com wrote: I would probably do this through the AMI, it should spew out the info you require. Timestamp when they entered the queue and timestamp when they get answered. On 10/25/2010 05:01 AM, Bruce B wrote: Anything on this guys? I am sure someone had the need to record the HOLD time or maybe it is already being recorded somewhere? Any thoughts are appreciated. Thanks, Bruce On Wed, Oct 20, 2010 at 3:30 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, We are using Queuemetrics but it doesn't Record the Hold Time as it's never logged on the queue_log file. However, when an agent or an extension presses HOLD button on their phone, asterisk does create an event for Music On Hold which is logged in the /var/log/asterisk/full. I want to record the total hold time for an extension and save it with an epoch time stamp. What is the best approach to this? read and parse /var/log/asterisk/full in a cron job every few seconds? Have a presistent PHP-AGI connection to check for hold time events? As much detail as possible on above approaches or other ideas are most appreciated. Thanks ---DISCLAIMER The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free. -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, October 25, 2010 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best way to recording the hold time for a Queue agent or an extension Thanks for the feedback. I don't need the Queue times but rather putting ON HOLD times. If you press the HOLD button on your SIP phone, Asterisk records the event Music On HOLD Playing and that is recorded in /var/log/asterisk/full. I want to harvest the ON HOLD time per phone SET. Thanks Again, the AMI would be your most likely help here. The information in /v/l/a/full is going to give you a start and stop time for MOH, but not in a format where you can easily tie it back to an extension. You will have a start moh and stop moh event in the AMI that is tied to an extension by the uniqueid. In PERL Weenie world, the way to process this is to pipe the AMI output where it is an input file keyed by the uniqueid. You can find some decent examples on voip-info.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] Chan variables for peer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Sunday, October 24, 2010 4:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Chan variables for peer
Hi all,
I used to configure each of my sip clients with a unique identifier via
setvar. These clients were all configured as friends.
However, now that I've got some Polycom phones, which MUST be peers, I am
unable to define this variable.
For example, this works:
[friend-client]
context = default
accountcode = pcc
type = friend
username = username
secret = ya,right
host = dynamic
nat = yes
canreinvite = no
callerid = User 155512345
mailbox = 155512...@customers,123456
setvar = id=123
However, if I change the type to peer, I am unable to get a value for
${id}.
Is this a known limitation, or am I doing something wrong? If this won't
work, is there a work-around?
--
Take care and have fun,
Mike Diehl.
#1 check the bug tracker
#2 you might have to change the syntax - it seems to me that setvar=id=123
is an accident waiting to happen (or maybe it did.) setvar=id=123 or setvar
= id=123 might be more appropriate.
--
_
-- 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] a2billing muting enter the phone number
But I don't want to delete the file! I just want to know where is the option for playing or disabling this message??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, October 25, 2010 10:01 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] a2billing muting enter the phone number -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH Sent: Saturday, October 23, 2010 7:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] a2billing muting enter the phone number How can I mute the message please enter the number you wish to call and press the # key in a2billing??? I tried use_dnid = YES but still I keep getting the message prompt... thanks the CLI is your friend here; when this prompt is playing, look at the CLI console and identify this file. It will be in /var/lib/asterisk/sounds. Replace it with beep.gsm. -- _ -- 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] a2billing muting enter the phone number
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH Sent: Monday, October 25, 2010 4:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] a2billing muting enter the phone number But I don't want to delete the file! I just want to know where is the option for playing or disabling this message??? That's an a2billing question, not an Asterisk one... -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
Thanks for the input. Would I have to process each call through a specific dial-plan for the AMI to be in charge of each call so that it can see the Unique ID of the channel and the Hold event? Because that seems like a lot of work. If AMI (I have no experience with it) allows me to open a socket and just read whatever comes through then it might work for me. Thanks On Mon, Oct 25, 2010 at 10:47 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Monday, October 25, 2010 9:32 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Best way to recording the hold time for a Queue agent or an extension Thanks for the feedback. I don't need the Queue times but rather putting ON HOLD times. If you press the HOLD button on your SIP phone, Asterisk records the event Music On HOLD Playing and that is recorded in /var/log/asterisk/full. I want to harvest the ON HOLD time per phone SET. Thanks Again, the AMI would be your most likely help here. The information in /v/l/a/full is going to give you a start and stop time for MOH, but not in a format where you can easily tie it back to an extension. You will have a “start moh” and “stop moh” event in the AMI that is tied to an extension by the uniqueid. In “PERL Weenie” world, the way to process this is to pipe the AMI output where it is an input file keyed by the uniqueid. You can find some decent examples on voip-info.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] B410P - BRI NT 100 Ohm terminator
2010/10/25 Shaun Ruffell sruff...@digium.com On 10/23/2010 07:07 AM, Olivier wrote: Hi, My set up is : Asterisk with B410P in NT mode -cat5 straight cable Another PBX in TE mode Is the 100 Ohm terminator you can find on B410P boards, necessary when connecting in NT mode to another PBX (set in TE mode) ? If you have it set up in a point-to-point mode, you do not need the termination. It is required when you are daisy chaining devices (like when using point-to-multipoint i.e. you are connecting several ISDN phones to one B410P in NT mode). OK ! Thanks : I forgot about this point-to-multipoint situation as I've still never met it anywhere. -- _ -- 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] Chan variables for peer
On Monday 25 October 2010 8:53:19 am Danny Nicholas wrote: #1 check the bug tracker #2 you might have to change the syntax - it seems to me that setvar=id=123 is an accident waiting to happen (or maybe it did.) setvar=id=123 or setvar = id=123 might be more appropriate. Thank you. I should be able to get it working from here. -- Take care and have fun, Mike Diehl. -- _ -- 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] How to properly re-configure dahdi
2010/10/25 Tzafrir Cohen tzafrir.co...@xorcom.com On Sat, Oct 23, 2010 at 02:43:26PM +0200, Olivier wrote: Hi, How to properly re-configure dahdi, when for instance I want to change from TE to NT mode ? I'm planning the following operations : /etc/init.d/asterisk stop /etc/init.d/dahdi stop rmmod dahdi rm /etc/asterisk/dahdi-channels.conf rm /etc/dahdi/system.conf rm /etc/dahdi/modules nano /etc/dahdi/genconf_parameters dahdi_genconf -v modules dahdi_genconf -v system dahdi_genconf -v chandahdi Am I missing something ? I'm not sure that dahdi_genconf can tell those changes, as the driver does not report them. The modules file has not changed, anyway. I did a couple of tests since I posted this note and now, I've got the feeling that some cards when NT mode is enabled, are not always detected (and configured by dahdi_genconf) as such. I don't have the time to dig a bit further now but I'll do ASAP and report here. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, October 25, 2010 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best way to recording the hold time for a Queue agent or an extension Thanks for the input. Would I have to process each call through a specific dial-plan for the AMI to be in charge of each call so that it can see the Unique ID of the channel and the Hold event? Because that seems like a lot of work. If AMI (I have no experience with it) allows me to open a socket and just read whatever comes through then it might work for me. In my experience, AMI is agnostic to how many calls it is handling. Therefore you would identify each call by getting the uniqueid from the answer event and assigning the remaining events to that call by matching (whether it would be 1, 100 or 1000 calls). -- _ -- 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] particular sip registry and outbound proxy
Hi, My asterisk's version is 1.6.0.26. I've couple sip providers and I've for new SIP provider I need define outbound proxy. Everything is ok in peer section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I need send SIP register messages also via outbound proxy. How to write SIP OUTBOUND call register statement and send this to proxy? If I define in general section this: outboundproxy=192.0.2.1 Works OK , but now Asterisk sends all SIP messages via this 192.0.2.1 proxy! So after changes other carrier stopped to work. Is it possible to write SIP registration statement to one provider and send these messages via outbound proxy? I mean to have multiple registration line and have different outboundproxy for each line. Thanks //Dante -- _ -- 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] Re : saturation of bandwidth because of HANGUP
Any news for this problem. Who has this problem --- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit : De: ALAEDDINE abbech alasup...@yahoo.fr Objet: saturation of bandwidth because of HANGUP À: asterisk-users@lists.digium.com Date: Jeudi 21 octobre 2010, 17h55 Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks -- _ -- 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] Re : thousands Hangup per second /saturation of bandwidth
Any news for this problem. Who has this problem Why you don't answer. --- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit : De: ALAEDDINE abbech alasup...@yahoo.fr Objet: thousands Hangup per second /saturation of bandwidth À: asterisk-users@lists.digium.com Date: Jeudi 21 octobre 2010, 11h42 Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks -- _ -- 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] Re : saturation of bandwidth because of HANGUP
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ALAEDDINE abbech Sent: Monday, October 25, 2010 10:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re : saturation of bandwidth because of HANGUP Any news for this problem. Who has this problem --- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit : De: ALAEDDINE abbech alasup...@yahoo.fr Objet: saturation of bandwidth because of HANGUP À: asterisk-users@lists.digium.com Date: Jeudi 21 octobre 2010, 17h55 Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks If we had this problem, either we would have posted a reply or would be too busy figuring it out ourselves. I personally dont believe Asterisk would send out 1000 hangups in 1 second (how would you monitor this? AMI output? /var/log/asterisk/full?) When you wonder if anybody is reading your post, check the archives to see if it actually got there. Im not going to be arrogant enough to tell you that Asterisk is a U.S. based audience (the posts I get indicate that there are a Large contingent of UK, Indian posters), but do keep in mind that lots of the posters (hopefully) work and post on a primarily 5 day workweek. -- _ -- 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] Re : thousands Hangup per second /saturation of bandwidth
Un-self-top-posting... --- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit : De: ALAEDDINE abbech alasup...@yahoo.fr Objet: thousands Hangup per second /saturation of bandwidth À: asterisk-users@lists.digium.com Date: Jeudi 21 octobre 2010, 11h42 Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks On Mon, 25 Oct 2010, ALAEDDINE abbech wrote: Any news for this problem. Who has this problem Why you don't answer. 0) This is a volunteer list. Nobody is obligated to answer. 1) Maybe nobody else has experienced this problem. 2) Maybe you failed to provide any information that would allow anybody to offer any suggestions of how to resolve your problem. Let's start with some simple details... a) What OS and version? b) What version of Asterisk? c) What technology is used for the failing call? I'm assuming SIP... d) What endpoint is involved? For example, Cisco 7960 with 8.3 firmware. e) What does your dialplan look like? Please use show dialplan so we can see what Asterisk sees. f) What does the Asterisk console output show after upping debug and verbose levels. g) Can the problem be replicated with a different endpoint? For example, a Zoiper Communicator 1.18.6898 softphone? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Re : thousands Hangup per second /saturation of bandwidth
You didn't attach some debug output that shows some work, and you didn't even tell us what asterisk version are you using, which scenario is on, etc. Don't expect people to run and answer right away with an inmediate solution to this. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center El 25/10/10 10:52, ALAEDDINE abbech escribió: Any news for this problem. Who has this problem Why you don't answer. --- En date de : *Jeu 21.10.10, ALAEDDINE abbech /alasup...@yahoo.fr/* a écrit : De: ALAEDDINE abbech alasup...@yahoo.fr Objet: thousands Hangup per second /saturation of bandwidth À: asterisk-users@lists.digium.com Date: Jeudi 21 octobre 2010, 11h42 Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks -- _ -- 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] Dial plan help
Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com Nile -- _ -- 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] Dial plan help
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon Sent: Monday, October 25, 2010 12:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial plan help Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com Nile I'll have to agree that VDP is a nice tool, but it is just that - a tool. If you don't know how the dialplan and commands work, it will eventually dig you into a hole you won't get out of. -- _ -- 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] Dial plan help
Ok Thanks Guys. Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nile Kaledon *Sent:* Monday, October 25, 2010 12:06 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Dial plan help Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com Nile I’ll have to agree that VDP is a nice tool, but it is just that – a tool. If you don’t know how the dialplan and commands work, it will eventually dig you into a hole you won’t get out of. -- _ -- 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] Dial plan help
Chapters 4, 5 and 6 is a good start. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote: Ok Thanks Guys. Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:aster... -- _ -- Bandwidth and Colo... -- _ -- 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] Pop-up for MS Outlook 2007 recommended
Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce -- _ -- 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] Pop-up for MS Outlook 2007 recommended
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, October 25, 2010 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pop-up for MS Outlook 2007 recommended Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce Not specifically what you are looking for, but it is very simple to use Apache/Ajax to make AMI links to launch calls from anywhere. I would invest 30-240 minutes into this method before bothering with the other stuff that is out there. Also, will make it easier when you eventually jump to 1.8/1.10. -- _ -- 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] Dial plan help
Un-top-posting... On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote: Ok Thanks Guys.Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, 25 Oct 2010, Zeeshan Zakaria wrote: Chapters 4, 5 and 6 is a good start. Yep. That's where I'd start if I didn't even know enough to ask questions using the correct terminology. I always skip the first 3 chapters in any technical book because I figure the authors put them in just to fill out their commitment to the publisher so he can charge more for the book -- even when the book is available for free. I figure, why learn the foundation of a new technology when there are always mailing lists manned by volunteers waiting at my beck and call -- my time is worth more than theirs. The one thing I can't figure out is why everybody keeps adding me to their MUA kill lists... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Dial plan help
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, October 25, 2010 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial plan help Un-top-posting... On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote: Ok Thanks Guys.Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, 25 Oct 2010, Zeeshan Zakaria wrote: Chapters 4, 5 and 6 is a good start. Yep. That's where I'd start if I didn't even know enough to ask questions using the correct terminology. I always skip the first 3 chapters in any technical book because I figure the authors put them in just to fill out their commitment to the publisher so he can charge more for the book -- even when the book is available for free. I figure, why learn the foundation of a new technology when there are always mailing lists manned by volunteers waiting at my beck and call -- my time is worth more than theirs. The one thing I can't figure out is why everybody keeps adding me to their MUA kill lists... Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 The things I could write here... We want instant gratification and want to drive the car without knowing anything except where the gas goes. There are plenty of Canned Asterisks for folks who don't want to bother with details like installation and dialplans. Is it easier to read 600 pages or 600 Flames? -- _ -- 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] Dial plan help
You may check these videos too: http://www.youtube.com/watch?v=H1j5OrgL1og http://www.youtube.com/watch?v=7kNYuqOrP3w I find it useful, although I use visual dial plan rather than hand coding the dial plan. Either way you need to understand at least basics of asterisk dial plan structure. Rayan On 10/25/2010 7:55 PM, Jigar Joshi wrote: Ok Thanks Guys. Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com wrote: *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nile Kaledon *Sent:* Monday, October 25, 2010 12:06 PM *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Dial plan help Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com http://codezone.apstel.com Nile I’ll have to agree that VDP is a nice tool, but it is just that – a tool. If you don’t know how the dialplan and commands work, it will eventually dig you into a hole you won’t get out of. -- _ -- 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] Pop-up for MS Outlook 2007 recommended
Great suggestion but unfortunately for this client a proven technology is needed and we don't mind paying a bit for it so once the time is available we might do this the way you suggested. Thanks On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Monday, October 25, 2010 1:14 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Pop-up for MS Outlook 2007 recommended Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce Not specifically what you are looking for, but it is very simple to use Apache/Ajax to make AMI links to launch calls from anywhere. I would invest 30-240 minutes into this method before bothering with the other stuff that is out there. Also, will make it easier when you eventually jump to 1.8/1.10. -- _ -- 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] Pop-up for MS Outlook 2007 recommended
I think there is a new version of Outcall, the pop up was pretty good, but the dialout wasn't ideal in Win 7 , and I believe thats fixed now with good integration with 2007 and 2010 http://code.google.com/p/outcall/ You can buy commercial options from Biocom - who make Outcall http://www.bicomsystems.com/products/C/P/319/288/ Cheers Duncan On 26/10/2010, at 8:24 AM, unsero...@aol.com wrote: Did you already check Bria? I have not tested it yet but it seems to be very powerful. Unfortunately there is no trial version available. If you will give it a try I would be very interested in your opinion. http://www.counterpath.com/bria-for-microsoft-outlook.html Oliver -Original Message- From: Bruce B bruceb...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Oct 25, 2010 9:10 pm Subject: Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended Great suggestion but unfortunately for this client a proven technology is needed and we don't mind paying a bit for it so once the time is available we might do this the way you suggested. Thanks On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, October 25, 2010 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pop-up for MS Outlook 2007 recommended Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce Not specifically what you are looking for, but it is very simple to use Apache/Ajax to make AMI links to launch calls from anywhere. I would invest 30-240 minutes into this method before bothering with the other stuff that is out there. Also, will make it easier when you eventually jump to 1.8/1.10. -- _ -- 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 -- _ -- 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] Extension Exists
Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551...@incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan -- _ -- 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] Asterisk 1.80-rc5
On Fri, 2010-10-22 at 11:16 +0200, Dave Cotton wrote: On 22/10/10 11:05, Hans Witvliet wrote: On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote: On 21/10/10 22:04, Hans Witvliet wrote: For suse there is a precompiled version on the OBS (vitsoft) Package search on the OBS shows nothing for 1.8.0 at all. Perhaps you know where it is hidden. Dave Cotton http://software.opensuse.org/search?q=asterisk18baseproject=openSUSE%3A11.3〈=enexclude_debug=true Trick is to DE-select the button for searching thrue home directories. When doing so, you find the maintainer of this package... OK thanks, I'd actually sorted that out, but this answer will help others. Regards Dave Cotton For those who might be interested... If possible i rather use maintream prebuild packages. As from now, they (asterisk180) are available for openSUSE_11.1, openSUSE_11.2, openSUSE_11.3, SLES10, SLES11, SLES11SP1 via: http://software.opensuse.org/search?q=asterisk180baseproject=openSUSE% 3A11.3lang=enexclude_filter=home%3Aexclude_debug=true enjoy it btw, if you want to rebuild them, the source rpm's with their spec files are there also. 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
Re: [asterisk-users] Configuring Asterisk behind a SIP Proxy
Hi,
I have the same issue. Did you solved it?
On Thu, Jun 18, 2009 at 5:21 PM, Brad Johnson bjohn...@ecessa.com wrote:
We are trying to configure Asterisk (version 1.6.1.0) with some SIP
phones behind a SIP Proxy/NAT device. The phones register properly to
Asterisk, and to get Asterisk to register properly to the external SIP
registrar we added this to the general section of sip.conf (the address
of the Asterisk system on the LAN is 192.168.30.5):
outboundproxy=192.168.30.10
register = myname:mysec...@my.provider.com/100
The problem we are facing is that it appears that the outboundproxy
value is being treated globally by Asterisk so it sends all SIP traffic,
including traffic to the phones, to the proxy. The behavior we want is
that all outbound traffic is sent to the proxy, but inbound SIP traffic
to the phones should be sent direct to the phones.
The result we see is that an inbound Invite is received by Asterisk and
then the Invite for the phone is sent by Asterisk to the outbound proxy.
This causes much confusion.
Can anyone please tell me how to configure Asterisk properly for working
behind a SIP Proxy?
Below you will find our configuration.
Thanks,
Brad
Here is the channel for our SIP provider:
[my_provider]
type=peer
host=my.provider.com
username=100-phone
secret=mysecret
context=incoming
canreinvite=no
qualify=300
insecure=port,invite
Here is a sample phone entry in sip.conf:
[100_phone]
type=friend
username=100-phone
secret=100secret
host=dynamic
context=internal
Here is the relevant part of extensions.conf:
[incoming]
exten = 100,1,Dial(SIP/100_phone,30)
exten = 100,n,Hangup()
[internal]
exten = _X.,1,Dial(SIP/my_provider/${EXTEN})
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards,
Giedrius
--
_
-- 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] Extension Exists
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 25, 2010 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extension Exists
Hi,
When a VOIP user dials an external number, the calls are routed through our
SIP provider.
Is there a simple way to check whether the DDI exists locally before
dialling out to the sip provider?
Something like GotoIfExists(5551...@incoming_calls)
Currently, I'm paying for all calls, regardless of whether they exist
locally.
All DDIs exist in the incoming_calls context.
Thanks
Dan
Here you go -
Exten = _X.,1,verbose(try local first)
Exten = _X.,n,Dial(SIP/${EXTEN},30,KkTt)
Exten = _X.,n,Dial(SIP/${ext...@provider,60,KkTt)
If the local exten does not exist, it should fall through immediately to the
provider call.
--
_
-- 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] Pop-up for MS Outlook 2007 recommended
Bria is a full SIP soft client. It works ok if you have a very good sound card and good wired headset. It is not a dialer application in the sense that you would dial your desk phone using it. Some of my clients love the Bria and some say the quality is poor. You must have a computer that can handle it the supporting sound and headsets. Bryant From: unsero...@aol.com Sent: Monday, October 25, 2010 3:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended Did you already check Bria? I have not tested it yet but it seems to be very powerful. Unfortunately there is no trial version available. If you will give it a try I would be very interested in your opinion. http://www.counterpath.com/bria-for-microsoft-outlook.html Oliver -Original Message- From: Bruce B bruceb...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Oct 25, 2010 9:10 pm Subject: Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended Great suggestion but unfortunately for this client a proven technology is needed and we don't mind paying a bit for it so once the time is available we might do this the way you suggested. Thanks On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, October 25, 2010 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pop-up for MS Outlook 2007 recommended Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce Not specifically what you are looking for, but it is very simple to use Apache/Ajax to make AMI links to launch calls from anywhere. I would invest 30-240 minutes into this method before bothering with the other stuff that is out there. Also, will make it easier when you eventually jump to 1.8/1.10. -- _ -- 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] Asterisk 1.80
For those who might be interested... If possible i rather use maintream prebuild packages. As from now, they (asterisk180) are available for openSUSE_11.1, openSUSE_11.2, openSUSE_11.3, SLES10, SLES11, SLES11SP1 via: http://software.opensuse.org/search?q=asterisk180baseproject=openSUSE% 3A11.3lang=enexclude_filter=home%3Aexclude_debug=true enjoy it btw, if you want to rebuild them, the source rpm's with their spec files are there also. Hans btw, I forgot to change the subject, on the obs it is not the RC anymore. repo/network:/telephony:/asterisk/SLE_11_SP1/src/asterisk180-1.8.0-83.1.src.rpm hw -- _ -- 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] Dial Plan Conf
Hi, I just downloaded your vdp file and it's working fine on my installation (Asterisk 1.4). Can you be more specific on the issue you experienced? Nile -- _ -- 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] particular sip registry and outbound proxy
Put the outboundproxy=192.0.2.1 under individual sip context not under the [general], it should work. CK On Mon, Oct 25, 2010 at 11:43 PM, sipbeast sipbe...@gmail.com wrote: Hi, My asterisk's version is 1.6.0.26. I've couple sip providers and I've for new SIP provider I need define outbound proxy. Everything is ok in peer section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I need send SIP register messages also via outbound proxy. How to write SIP OUTBOUND call register statement and send this to proxy? If I define in general section this: outboundproxy=192.0.2.1 Works OK , but now Asterisk sends all SIP messages via this 192.0.2.1 proxy! So after changes other carrier stopped to work. Is it possible to write SIP registration statement to one provider and send these messages via outbound proxy? I mean to have multiple registration line and have different outboundproxy for each line. Thanks //Dante -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Monday, October 25, 2010 07:30:22 am Stephen Reese wrote: Does the AGI have to be used? In this example http://www.davidvossel.com/?p=28 I see mention of a script, but not in this one: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ I believe I missing the connection in how the whole process actually works therefore making troubleshooting a little difficult. I was hoping with the release of 1.6.0 there wouldn't be a lot of bandage work to get it to play nicely with Google Voice. Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc., for outbound calls, it acts basically like a fancy click-to-call application. So... You need Asterisk to login into GV, and initiate the call. GV will dial the number you tell it to, then connect it to one of your GV numbers. In my case, the AGI is what connects to GV and initiates the call. GV, then dials the number I told it to dial, then connects it with my ipKall number (which I have as one of my GV numbers). In Asterisk, the outbound call runs the AGI and places the channel in the DB, then waits for an incoming call via my inbound ipKall trunk. Once the ipKall comes into Asterisk, the Bridge command is used to bridge the original (with the matching DB entry) call-- the call that is coming in from GV through ipKall. I suppose you don't need that AGI and could probably do this using Curl in the dialplan. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Mobile Phones and Asterisk
Hi, Is the dev_state can also be used to track a mobile phone's status via SIP? I tried it on several phones(nokia, samsung) but it returns NOANSWER but i can hear a beep beep beep sound indicating that it is currently busy. regards, RYAN ICASIANO -- _ -- 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] Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
The suggestions did fix the problem. Thank you Shaun and Paul for the help. Regards, Jared On Fri, Oct 15, 2010 at 4:48 PM, Jared Geiger ja...@compuwizz.net wrote: I haven't heard if this fixed it yet. However I was seeing the echo cancelers loaded before so I never realized I'd have to do this. Its a FreePBX install also so I checked all the include files and didn't see a reference to these values anywhere. Thanks everyone for the input, I should know soon if it is the fix. ~Jared On Fri, Oct 15, 2010 at 3:56 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger compuw...@gmail.com wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I did an update on a server last year, had the same problem. I needed to explicitly set echocancel=yes in my configs, before 1.6 it was enabled by default. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Extension Exists
On 10-10-25 04:21 PM, Dan Journo wrote: Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551...@incoming_calls) Well this is really an implementation question. If your data was in a database you could use func_odbc to check if the DID was local. You can check with VALID_EXTEN() to see if a particular extension exists locally. That's check the databse, so if you have a context that contains a list of your local DIDs you can check with that function. If the DIDs are available as a list on a webpage you can use func_curl. Using the DB_EXISTS() function could be used if storing in the Asterisk database. Those are some options. Leif. -- _ -- 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] ISDN SS7
Are these solutions reliable and stable ?. Have you used these solutions in production ? What about its quality ? From: Tzafrir Cohen tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Sent: Tue, October 26, 2010 3:12:21 AM Subject: Re: [asterisk-users] ISDN SS7 On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote: SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn't support SS7 natively although I believe there are one or more add-on vendors. The vendors of addons such as http://svn.asterisk.org/svn/libss7 and http://svn.asterisk.org/svn/asterisk/trunk/channels/sig_ss7.c ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] ISDN SS7
I'm planning to use SGM with Asterisk, it is a commercial product. What is the different between SGM and libs77 and chan_ss7 ? Should I use SGM ? From: Tzafrir Cohen tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Sent: Tue, October 26, 2010 3:12:21 AM Subject: Re: [asterisk-users] ISDN SS7 On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote: SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn't support SS7 natively although I believe there are one or more add-on vendors. The vendors of addons such as http://svn.asterisk.org/svn/libss7 and http://svn.asterisk.org/svn/asterisk/trunk/channels/sig_ss7.c ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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
