Re: [asterisk-users] Call Transfer Fails - Not a Valid Extension
The file /var/log/asterisk/full will contain helpful log messages that show how Asterisk is internally handling the call. It may be necessary to increase the verbosity of the log to get more details however. From the linux command line, you can follow these steps to get a copy of the relevant messages: # asterisk -rx core set verbose 5 # cat /var/log/asterisk/full mylogfile (perform a transfer that fails with the message now, then press CTRL-C to cancel the above command) The mylogfile will have the log entries necessary to understand what happened, although it may also require an understanding of the FreePBX dialplan to interpret it. If you can post your log file (recommend using a pastebin rather than emailing the whole thing) it should be fairly easy to spot the problem and advise you how to fix it. On Sun, Sep 7, 2014 at 10:55 PM, Phil Ledon ple...@lodgetech.com wrote: We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones. All transfers are failing from any source to any extension with the message “that is not a valid extension”. Does anyone have any ideas about where to begin looking for the source of that error? *Phil Ledon* -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Call transfer problem.
You have to use attendant transfer, not blind. - A calls B - B answers on line 1 (button 1) - B has to use line 2 (push button 2) to call C, C sees call coming from B, the same does asterisk - while having line 2 active, he pushes button transfer followed by button line 1 - A speaks with C On Mon, Feb 24, 2014 at 7:45 PM, Mike Diehl mdiehlena...@gmail.com wrote: I'm sorry, I should have mentioned that he's doing a phone-based transfer, not an asterisk-based transfer. Mike. On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly d...@donkelly.biz wrote: Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- 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
Re: [asterisk-users] Call transfer problem.
I'm sorry, I should have mentioned that he's doing a phone-based transfer, not an asterisk-based transfer. Mike. On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly d...@donkelly.biz wrote: Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- 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] Call transfer problem.
Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer question
Hi faheem, You can do this: ACTION: Redirect Channel: Channel ID Context: Context Exten: Exten Priority: Priority Regards, Qasim On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem faheem2...@gmail.comwrote: Hi, is possible that two sip extensions: user-1 and user-2 are connected and I want to transfer the call from user-1 to a third user user-3. I know it is possible through feature keys mapping in features.conf, but I want to do this through AMI or Asterisk CLI Commands? Please suggest if possible? Thank you! Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
Thanks everyone. I was using the Tt flag but in the wrong place in the dial application. Cheers On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima takehiro.dream...@gmail.com wrote: Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.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] Call Transfer not working
Hi. Maybe you forgotten specify to allow the transferring a call. Try with tT options in Dial() in extensions.conf. // I don't know what's difference t and T. -- Takehiro Matsushima takehiro.dream...@gmail.com 2012/4/7 Rizwan Hisham rizwanhas...@gmail.com: Hi All, I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with dtmf setting rfc2833 and inband. I have also enabled blind and attended transfer features in features.conf but still call transfers dont work. I have setup transfer feature in past but i dont think i am missing anything this time. I just dont have any clue why its not working. I have tried using ATAs and softphones but cant make it to work. Can anyone help? Am I missing anything? features show output: === Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # #1 Attended Transfer *2 One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor == -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.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] call transfer
On Tue, 2010-02-16 at 17:25 +0530, cool dude wrote: call transfer call transfer from reception to other extensions. Question: Details of Extensions Reception - 2000 Sales - 2001 Accounts - 2002 any call comes it should be received by extenion 2000, n if person wants to talk to Sales, receptionist should put the caller on hold than connect to Sales i.e exten 2001, while on hold the caller should hear music on hold,now sale exten can take his call n talk to it.same with Accounts ext 2002. vi /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=reception secret=1234 host=dynamic [2001] type=friend context=sales secret=1234 host=dynamic [2002] type=friend context=accounts secret=1234 host=dynamic ~ ## vi /etc/asterisk/extension.conf [from-zaptel] exten = s,1,wait(2) exten = s,n,Dial(SIP/2000,20) what to next __ What have you tried? Which links/mans have you read to set up music on hold? Are any of them wrong at all? -- _ -- 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
Re: [asterisk-users] call transfer
Tuesday, February 16, 2010, 12:55:12 PM, cool wrote: call comes it should be received by extenion 2000, n if person wants to talk to Sales, receptionist should put the caller on hold than connect to Sales i.e exten 2001, while on hold the caller should hear music on hold,now sale exten can take his call n talk to it.same with Accounts ext 2002. ... what to next To have call transfer in your asterisk setup, YOU need to read some documentation. Start here: http://www.voip-info.org -- Best regards, Gergomailto:csi...@gmail.com -- _ -- 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
Re: [asterisk-users] call transfer using DTMF
On 15/7/09 3:07 PM, Michael wrote: Is there a way to transfer a call, while in the middle of the call, using DTMF? Yep, just pass the t or T options to the dial command and set it up in /etc/asterisk/features.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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
Re: [asterisk-users] call transfer using DTMF
Yes, In the features.conf under featuremap you need the blindtransfer un-commented blindxfer = ## Then in your extensions.conf you need to have at least a capital T exten = example,1,Dial(ZAP/4/12345,,T) Then during the call you can press ## and asterisk will say transfer. Then dial in the extension you want to transfer too. Thank you, Brad Finberg - Original Message - From: Michael as...@nettrust.co.nz To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Date: Tuesday, July 14 2009 11:07 PM Subject: [asterisk-users] call transfer using DTMF Is there a way to transfer a call, while in the middle of the call, using DTMF? ___ -- 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___ -- 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
Re: [asterisk-users] call transfer in CDR
On Thu, Jan 15, 2009 at 4:09 AM, Rilawich Ango maillist...@gmail.com wrote: Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango You may want to read this thread. http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html Regards, Greyman. ___ -- 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
Re: [asterisk-users] Call transfer using agi
You could simply have it Dial() to wherever it needs to go at the end of the script. 2009/1/6 Rajkumar S rajkum...@gmail.com Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. I have an agi to change password and can transfer call to agi, but I do not know how to transfer the call back to agent from agi. So basically how can an agi transfer a call to an extension? Thanks and regards, raj ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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
Re: [asterisk-users] Call transfer over IAX trunk
Ciao Noah, What flags do you have in your Dial() statement? If you want both parties to be able to transfer with the features.conf transfer, you need to have 'Tt' in your dial statement, like this: Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt) Bingo. That was the problem. Thanks a lot, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer over IAX trunk
Hi Andrea - I have two asterisk servers, an IAX trunk between and some SIP users registered to each server. The scenario is this: user A, registered to PBX 1, calls user B, registered to PBX 2. Then A wants to transfer the call using the features.conf method (in my case, **), but is unable to do this. What flags do you have in your Dial() statement? If you want both parties to be able to transfer with the features.conf transfer, you need to have 'Tt' in your dial statement, like this: Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt) - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer
On Sat, Jun 7, 2008 at 8:24 AM, Theodore Patsiouras [EMAIL PROTECTED] wrote: If my secretary or anyone else picks up the call when the line is transferred in my ext then I have the internal caller ID. Can I have somehow the External callerID? Look at the channel variables that contain the callerid information. You can assign the incoming callerid to the one that makes the call to your local extension to do what you wish. ___ -- 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
Re: [asterisk-users] call transfer detection in dial plan
On 9/13/07, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, In default, we can use # to transfer the call. I want to know how I can know the user presse # to transfer the call in dial plan. ango Set TRANSFER_CONTEXT or GOTO_ON_BLINDXFER variable (depending on * version) before Dial(). I just wrote more explaining mail to list. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] call transfer not working
check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general settings. On 7/4/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]---[Mediant2k][Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task regards -- Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --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 Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com ___ --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
Re: [asterisk-users] Call transfer in asterisk
On Monday 02 July 2007 01:45:44 pm satish patel wrote: dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://www.voip-info.org/wiki/index.php?page=PBX+CallTransfer -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.com ___ --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
Re: [asterisk-users] Call transfer in asterisk
satish patel wrote: dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel And this: http://www.voip-info.org/wiki-Asterisk+config+features.conf Warm Regards, Lee ___ --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
Re: [asterisk-users] Call transfer in asterisk
Dear all i have read that document but dont understand about function i have include featuremap in extension.conf [mysip] include = featuremap and reload extention.conf i got this error *CLI extensions reload Jul 2 19:23:04 WARNING[16320]: pbx.c:6444 ast_context_verify_includes: Context 'mysip' tries includes nonexistent context 'featuremap' *CLI also i have chenged in feature.conf [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer why my inculde function not working properly Lee Jenkins [EMAIL PROTECTED] wrote: satish patel wrote: dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel And this: http://www.voip-info.org/wiki-Asterisk+config+features.conf Warm Regards, Lee ___ --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 - Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. ___ --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
Re: [asterisk-users] Call transfer feature
satish patel wrote: Dear ALL I want to transfer call from one phone 2 another phone so this is asterisk feature or SIP Phone feature or endpoint feature how can i transfer phone call from to another phone Rgd Satish patel Check out this page: http://www.voip-info.org/wiki-Asterisk+config+features.conf -- Warm Regards, Lee ___ --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
RE: [asterisk-users] CALL TRANSFER
Your dial string must have either the t or T option set. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of omar parihuana Sent: Friday, December 01, 2006 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CALL TRANSFER Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALL TRANSFER
Thanks!!! I forget Tt option! (too basis!!) On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote: Your dial string must have either the t or T option set. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *omar parihuana *Sent:* Friday, December 01, 2006 9:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CALL TRANSFER Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CALL TRANSFER
Be careful, if you set both T and t you might be allowing the wrong party to transfer the call! In MOST cases you would want T or t, not T and t, although there are some cases where you might want both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of omar parihuana Sent: Friday, December 01, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CALL TRANSFER Thanks!!! I forget Tt option! (too basis!!) On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote: Your dial string must have either the t or T option set. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of omar parihuana Sent: Friday, December 01, 2006 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CALL TRANSFER Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are working fine. Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in accordance to voip-info.org http://voip-info.org/ but the transfer doesn't work! Please if you can provide me some examples will be very appreciate. Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe http://www.usysnet.com.pe/ ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer issues
My guess is I stumped everyone ;)
Anyway, I rolled back asterisk to 1.2.9.1 (same for libpri and zaptel
back one release) and transfers were working again. Now I'm still quite
new to asterisks, I know enough to hold my own, but not enough to know
the full inter workings of it. But here is my thought:
Caller A calls in and talks to Employee B. B wants to transfer to C.
Asterisk sets up the bridge between B and C. B completes the transfer.
Now A and C are connected but there is no audio stream. If C or A puts
the other on hold, and then resumes the call, audio is restored.
By that I would say placing them on hold clears a flag or updates one to
connect the audio stream? Or am I way off on this assumption? Also if
this sounds like a possible bug, what information do I need to include,
or is good to include, when submitting bugs?
Thanks,
Kevin
Kevin Smith wrote:
Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers were you transfer the call before the person
picks up works.
3. If the person the call is being transferred to answers and then the
transfer completes, the audio drops.
I noticed in the CLI the following (I replaced the number with XXX's)
-- Attempting native bridge of SIP/989XXX-b76167c8 and
SIP/989XXX-08f956b8
== Parsing '/etc/asterisk/manager.conf': Found
-- Stopped music on hold on Zap/2-1
== Spawn extension (Mercury-Directory-Dialer, 989XXX, 8) exited
non-zero on 'SIP/989XXX-b76167c8ZOMBIE'
-- Executing Hangup(SIP/989XXX-b76167c8ZOMBIE, ) in new
stack
== Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero
on 'SIP/989XXX-b76167c8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back
from 64.7.177.103
Now what I noticed is that once the transfer is done, I'm still
connected the the person that called me to do an attended transfer.
However, if I hang up the phone, the call drops. If I place the call
on hold and take them off hold, audio is resumed and everything works
normally.
Here is the conf information
exten = s,1,SetCallerID(${ARG1})
exten = s,n,Set(DST_EXT_NUM=${ARG2})
exten = s,n,gotoif,$[${ARG2}=989XX]?TIME:GOON ;Add test if
hours is the basis for voice mail
exten = s,n(GOON),AGI(VoiceMail.php) ;Test for phone status
exten = s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
exten = s,n,Dial(SIP/${ARG2},25)
...VoiceMail choice
exten = h,1,HangUp()
Where I have VoiceMail choice it takes the variables from the AGI
script and decides which voice message to play. But the problem is
happening before that occurs so I don't think it has anything to do
with the problem.
Any ideas to what could be the cause or how to correct it? SIP version
or does the new asterisk build have any new features enabled by
default that the older build would not? Any suggestions or thoughts
would be greatly helpful.
Kevin
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Re: [Asterisk-Users] call transfer to external phone number
Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe I know that with polycom I was able to do this. Not by using the # sign but by hitting the transfer button and then entering the persons number and pressing transfer. Please post your extensions.conf __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the phone transfering has an exten declared for that number. On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote: Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call transfer to external phone number
From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Monday, April 03, 2006 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call transfer to external phone number Yes, as long as the context that the phone transfering has an exten declared for that number. Does Asterisk make any distinction between an internal number and an external number? I'm inclined to think it might be some kind of timeout issue. And I've got the proof: From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can transfer a call to any extension, including the lng extension required for dialing an external number (ie: #0X). Unfortunatelly that's the ONLY phone I can do that from! I can't do it from XLite softphone and I can't do it from analog phones connected to a Linksys PAP2. For the phones that are unable to transfer to external numbers I've got alias extensions defined (basic, 3 digit extensions). On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote: Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer to external phone number
Asteirsk has got no clue what's internal and what's not, it's the context that decide what numbers are available for a user. In your case more info is needed to troubleshoot it. On 4/3/06, Cosmin Prund [EMAIL PROTECTED] wrote: From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Monday, April 03, 2006 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call transfer to external phone number Yes, as long as the context that the phone transfering has an exten declared for that number. Does Asterisk make any distinction between an internal number and an external number? I'm inclined to think it might be some kind of timeout issue. And I've got the proof: From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can transfer a call to any extension, including the lng extension required for dialing an external number (ie: #0X). Unfortunatelly that's the ONLY phone I can do that from! I can't do it from XLite softphone and I can't do it from analog phones connected to a Linksys PAP2. For the phones that are unable to transfer to external numbers I've got alias extensions defined (basic, 3 digit extensions). On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote: Hi! Is it possible to transfer a call to an external phone instead of transferring the call to internal phone? (I'm sorry for my bad english, I hope you understand) When, during a call, I digit #123, the call is transferred to internal extension 123, but if I digit #external_phone_number, it tells me that it's impossible. Any idea? Thanks a lot! Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer to external phone number
SIP transfers happen out of band, so the context is the sip phone's
context noted in sip.conf.
For Inbound and outbound (ie Dial application), the context is the entry
point in the dial plan. If you need features.conf transfers to work in
a specific context you need to set the __TRANSFER_CONTEXT variable
before the Dial application so asterisk knows what context to look for
extensions.
The relevant wiki page:
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
example:
exten = 101,1,set(__TRANSFER_CONTEXT=vm-internal)
exten = 101,n,Macro(superdial,SIP/vm-ext1SIP/outsip/9995551212,15,tr,
,pstn,2,${CALLERIDNAME},${CALLERIDNUM},pstn,[EMAIL PROTECTED])
So .. in this instance, when we outdial the cellphone (9995551212) with
the 't' option, we support transfers. If we don't set the transfer
context as above when the # key is hit. Asterisk is looking in the
[inbound] context because that is where extension 101 was dialed from.
But ... exten 101 doesn't want those available extensions, they want the
same set of extensions they have at their sip phone so they can transfer
to voicemail and so on.
Since our outbound pattern dials to SIP/outsip also exist in
[vm-internal] .. calls can be transferred out to PSTN numbers.
in any case.. this is how I got it to work. :)
Cosmin Prund wrote:
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, April 03, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the phone transfering has an exten
declared for that number.
Does Asterisk make any distinction between an internal number and an
external number? I'm inclined to think it might be some kind of timeout
issue. And I've got the proof:
From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can
transfer a call to any extension, including the lng extension required
for dialing an external number (ie: #0X). Unfortunatelly that's the
ONLY phone I can do that from! I can't do it from XLite softphone and I
can't do it from analog phones connected to a Linksys PAP2.
For the phones that are unable to transfer to external numbers I've got
alias extensions defined (basic, 3 digit extensions).
On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote:
Hi!
Is it possible to transfer a call to an external phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you understand)
When, during a call, I digit #123, the call is transferred to internal
extension 123,
but if I digit #external_phone_number, it tells me that it's
impossible.
Any idea?
Thanks a lot!
Giuseppe
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Re: [Asterisk-Users] Call Transfer
I would use asterisk's built in blind or attended transfer features. This way the system is based around dtmf and the users aren't tied to a specific model of phone to accomodate future upgrades. In order to do this I would recommend editing features.conf so blindxfer = ** instead of *. A single * for transfer makes it real difficult to use banking and other IVRs that ask you to press #. Then, in each necessary Dial cmd of your dialplan, make sure there is a t or a T in the options to enable either the called or calling users to initiate the transfer. More details about the Dial command: http://www.voip-info.org/wiki-Asterisk+cmd+Dial Moj Dave Morrow wrote: Can anyone point me in the right direction. My users (all using Sipura SPA-841 phones) need the ability to transfer a call to another number. How can I setup a dial plan to do this? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _http://www.autodatasolutions.com_ NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at_ [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer
I got this to work by editing the line
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
Do you know of anyway to set it up through AMP, so it works with all calls?
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000
Michael Sampson wrote:
I'm not sure how this is suppose to work. But I want to be able to
call people from a SIP phone and transfer them into a conference room.
If I call another extension that is a SIP phone I can hit # and then
enter the conference room number. If I call from the PSTN to the SIP
extension phone I can transfer by hitting # too. But if I call from
the SIP phone extension to a PSTN number it doesn't do anything when I
hit the #. I'm using [EMAIL PROTECTED] and under general settings I have
tTrwW for Asterisk Dial Command Settings.
Can you call through a Zap trunk from a SIP phone and do a call transfer?
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Re: [Asterisk-Users] Call transfer with voicemail password
http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords Cheers 2005/12/1, Joe Pukepail [EMAIL PROTECTED]: Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet. On 11/30/05, Benjamin Lenard [EMAIL PROTECTED] wrote: Hi, I'm trying to have an extension ring my SIP phone then try my cell phone. I can transfer the call fine to the cell but I want it to ask for a pin , voicemail pin, before transferring the call. This is so if my cell's voicemail answers , the call doesn't transfer to it. Any ideas? Thanks, Ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer with voicemail password
Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet. On 11/30/05, Benjamin Lenard [EMAIL PROTECTED] wrote: Hi,I'm trying to have an extension ring my SIP phone then try my cellphone.I can transfer the call fine to the cell but I want it to ask for a pin , voicemail pin, before transferring the call.This is so if my cell's voicemail answers , the call doesn't transferto it.Any ideas?Thanks,Ben___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer and pick chan_h323
AFAIK there were some known issues preventing call transfer from H323
terminals, at least with Innovaphone ones.
Yours
l.
On Fri, 18 Nov 2005 17:45:39 +0100, Santosh Rao
[EMAIL PROTECTED] wrote:
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls
can be made and recieved to and from extensions.
How to implement call transfer and call pickup. when using asterisk
1.0.x dtmf=inband registers and sends dtmf but with asterisk 1.2 and
ooh323 it does not.. is this a known issue ? While google heard tht
there was a issue with chan_h323.so would not send inband so tried to
install chan_0h323.so but but.. asterisk refuses to start with
chan_oh323 it says Unregistered channel type 'Modem'
my basic requirements are h323 , call pickup and call transfer? below
attached are the configurations files tht we are using currently ...
thanking for all your support ..
Extensions.conf:-
[testing]
exten = _7.,1,Pickup({66}:[EMAIL PROTECTED])
exten = 666,1,Dial(H323/192.168.1.194,100,Ttr)
exten = 667,1,Dial(H323/192.168.1.195,100,Ttr)
exten = 668,1,Dial(H323/192.168.1.196,100,Ttr)
exten = 669,1,Dial(H323/192.168.1.192,100,Ttr)
H323.conf:-
[general]
port = 1720
bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for
this machine
disallow=all
allow=ulaw
allow=alaw
;dtmfmode=auto
dtmfmode=inband
gatekeeper = DISABLE
context=testing
[vivek]
type=friend
host=192.168.1.194
context=testing
Callgroup=1
pickupgroup=1-9,13
[santosh]
type=friend
host=192.168.1.195
context=testing
Callgroup=1
pickupgroup=1-9,13
[binu]
type=friend
host=192.168.1.196
context=testing
Callgroup=1
pickupgroup=1-9,13
[test1]
type=friend
host=192.168.1.192
context=testing
Callgroup=1
pickupgroup=1-9,13
Features.conf:-
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
pickupex = *8
[featuremap]
blindxfer = #1 ; Blind transfer
atxfer = *2 ; Attended transfer
I haven't lost my mind; it's backed up on
tape somewhere.
Santosh Rao
Trikon Electronics Pvt Ltd
--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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Re: [Asterisk-Users] Call Transfer problems-am I missing something?
Hi,
Thanks for the clarification. I had seen that the two options
existed, but the docs for the dial() command didn't state the
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension. Does
anyone know why the calling party can't initiate the transfer? am I
missing something?
Yes. The ,t in the Dial() options is for callee, the T is for
caller. ,tT is for both.
Ciao,
David A. Bandel
--
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Re: [Asterisk-Users] Call Transfer problems-am I missing something?
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension. Does
anyone know why the calling party can't initiate the transfer? am I
missing something?
Yes. The ,t in the Dial() options is for callee, the T is for
caller. ,tT is for both.
Ciao,
David A. Bandel
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Re: [Asterisk-Users] Call Transfer problems-am I missing something?
David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension. Does
anyone know why the calling party can't initiate the transfer? am I
missing something?
Yes. The ,t in the Dial() options is for callee, the T is for
caller. ,tT is for both.
As is documented in show application dial in the Asterisk CLI.
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Re: [Asterisk-Users] Call Transfer
try # and then dial the extension.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote: Hello,I have my [EMAIL PROTECTED] working beautifully for basic call function. So now Iam testing extended functions for my office users and am hitting a wall.I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone andaccording to the directions, Isimply select Hold enter ext hit Fwd.However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * toallow call transfer to work? I am using an inbound trunk from Teliax- nocards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapateltrunks/configurations.I have also seen a lot of information for callforwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help youwith that, let me transfer you and then be able to do it. Since thishappens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phoneitself.Thanks-R___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205 Phone: 978-203-3848 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried yoursuggestion. No go. I have 3 of the Eezee phones andcall transferdoesn'twork on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had noreference to blindxfer or atxfer. I added them so my feature.conf nowlooks like this:transferdigittimeout = 3; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer iscomplete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension.Default is *8 ;featuredigittimeout = 500; Max time (ms) between digits for ; feature activation.Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1; One Touch Record atxfer = *2I rebooted my * server but still no go. Are theredependenciesI amnot aware of? Should [featuremap] be referenced elsewhere as well? I amworking with * CVS 1.0.9 and have found an article on wiki that supportfor call transfer was added in 1.2.Are there other places I need tohack for this functionality?Thanks,-RTom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, Isimply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations.I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
It is set to rfc2833. Tom Vile wrote: maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Call Transfer
I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote:It is set to rfc2833.Tom Vile wrote: maybe its not setting the DTMF tones properly.What do you have setup for the phone and extensions.Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones andcall transfer doesn'twork on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500; Max time (ms) between digits for; feature activation.Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are theredependenciesI am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2.Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, Isimply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations.I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design
Re: [Asterisk-Users] Call Transfer
Blind transfer should work fine #. Can you dial into Voicemail and enter your password succesfully?On 10/20/05, BJ Weschke [EMAIL PROTECTED] wrote:I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote: It is set to rfc2833.Tom Vile wrote: maybe its not setting the DTMF tones properly.What do you have setup for the phone and extensions.Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones andcall transfer doesn'twork on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500; Max time (ms) between digits for; feature activation.Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are theredependenciesI am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2.Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, Isimply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server.I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations.I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com mailto: Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Call Transfer
Yes, I can dial *97 for VM and check messages. When I select # during a call it does nothing though. I tried inband for DTMF but that didnt work. Am going to run debug mode ( first I have to figure out how :) ) and I will let you know what I find out. Thanks so far, R Tom Vile wrote: Blind transfer should work fine #. Can you dial into Voicemail and enter your password succesfully? On 10/20/05, *BJ Weschke* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It is set to rfc2833. Tom Vile wrote: maybe its not setting the DTMF tones properly. What do you have setup for the phone and extensions. Usually its rfc2833 but could be inband. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have the phone specific directions to transfer calls, but I tried your suggestion. No go. I have 3 of the Eezee phones and call transfer doesn't work on any of them, so I really don't think it is hardware related. I think the problem may be with my feature.conf which had no reference to blindxfer or atxfer. I added them so my feature.conf now looks like this: transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #; Blind transfer disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 I rebooted my * server but still no go. Are there dependencies I am not aware of? Should [featuremap] be referenced elsewhere as well? I am working with * CVS 1.0.9 and have found an article on wiki that support for call transfer was added in 1.2. Are there other places I need to hack for this functionality? Thanks, -R Tom Vile wrote: try # and then dial the extension. On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone and according to the directions, I simply select Hold enter ext hit Fwd. However when I press the button all I do is annoy the caller with loud button punching sounds. Does something need to be configured in * to allow call transfer to work? I am using an inbound trunk from Teliax- no cards, just a T1 direct to my * server. I have found transfer functions for zapatel- but as I said I am just using the T1 and have no zapatel trunks/configurations. I have also seen a lot of information for call forwarding but that sets up a permanent forward function to a specific extension. I just want to be able to say One moment, Mike can help you with that, let me transfer you and then be able to do it. Since this happens with all my AT-320 phones, I don't think it is hardware related and there is no mention of call transfer configuration for the phone itself. Thanks -R ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com
Re: [Asterisk-Users] Call Transfer
He is what happens from the time the extension is selected from the time
the digital receptionist answers until I hangup. I watched the logs as
I was pushing all sorts of transfer button possibilities and nothing. It
just stayed at 'ooh, voice format changed to 4' Which, while humorous
tells me nothing except that my phone is not able to communicate with
the sever at all from the time the call is put through until the call is
done.
Oct 20 15:33:45 VERBOSE[2909]: -- Executing
Dial(IAX2/[EMAIL PROTECTED]/4, IAX2/7878|15|tr) in new stack
Oct 20 15:33:45 DEBUG[2909]: SIMPLE DIAL (NO URL)
Oct 20 15:33:45 VERBOSE[2909]: -- Called 7878
Oct 20 15:33:45 VERBOSE[2909]: -- Call accepted by xxx.xxx.xxx.xxx
(format ulaw)
Oct 20 15:33:45 VERBOSE[2909]: -- Format for call is ulaw
Oct 20 15:33:45 VERBOSE[2909]: -- IAX2/7878/8 is ringing
Oct 20 15:33:50 VERBOSE[2909]: -- IAX2/7878/8 answered
IAX2/[EMAIL PROTECTED]/4
Oct 20 15:33:50 VERBOSE[2909]: -- Attempting native bridge of
IAX2/[EMAIL PROTECTED]/4 and IAX2/7878/8
Oct 20 15:33:50 DEBUG[2909]: Ooh, voice format changed to 4
Here is the extension config for 7878:
exten = 7878,1,Macro(exten-vm,[EMAIL PROTECTED],7878)
And this is the config for aah_1 ( our digital receptionist)
[aa_1]
include = aa_1-custom
exten = 1,1,Goto(,s,1);
exten = fax,1,Goto(ext-fax,in_fax,1);
exten = h,1,Hangup();
exten = i,1,Playback(invalid);
exten = i,2,Goto(s,7);
include = ext-local
include = app-messagecenter
include = app-directory
exten = s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4);
exten = s,2,Answer();
exten = s,3,Wait(1);
exten = s,4,SetVar(DIR-CONTEXT=default);
exten = s,5,DigitTimeout(3); Basic
exten = s,6,ResponseTimeout(7);
exten = s,7,Background(custom/aa_1);
Thanks, once again,
-R
Rhonda Herron wrote:
Yes, I can dial *97 for VM and check messages. When I select # during
a call it does nothing though. I tried inband for DTMF but that didnt
work. Am going to run debug mode ( first I have to figure out how :) )
and I will let you know what I find out.
Thanks so far,
R
Tom Vile wrote:
Blind transfer should work fine #. Can you dial into Voicemail and
enter your password succesfully?
On 10/20/05, *BJ Weschke* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I'm not sure the txfer functionality is in the 1.0.X branch. I'm
pretty sure you will need HEAD or the 1.2 betas.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
It is set to rfc2833.
Tom Vile wrote:
maybe its not setting the DTMF tones properly. What do you
have setup
for the phone and extensions. Usually its rfc2833 but could
be inband.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I have the phone specific directions to transfer calls,
but I
tried your
suggestion. No go. I have 3 of the Eezee phones and call
transfer
doesn't work on any of them, so I really don't think it
is hardware
related. I think the problem may be with my feature.conf
which had no
reference to blindxfer or atxfer. I added them so my
feature.conf now
looks like this:
transferdigittimeout = 3 ; Number of seconds to
wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to
the parked
caller
; when someone dials a
parked call
xfersound = beep ; to indicate an attended
transfer is
complete
xferfailsound = beeperr; to indicate a failed
transfer
;adsipark = yes ; if you want ADSI parking
announcements
;pickupexten = *8 ; Configure the pickup
extension.
Default is *8
;featuredigittimeout = 500 ; Max time (ms) between
digits for
; feature
activation. Default is 500
[featuremap]
blindxfer = #; Blind transfer
disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
atxfer = *2
I rebooted my * server but still no go. Are
there dependencies I am
not aware of? Should [featuremap] be referenced elsewhere
as well?
I am
working with * CVS 1.0.9 and have found an article on
wiki that
support
for call transfer was added in 1.2. Are there other
places I need to
hack for this functionality?
Thanks,
-R
Tom Vile wrote:
try # and then dial the extension.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED]
Re: [Asterisk-Users] Call transfer - atxfer
Are you using 1.0.x? DTMF Attended Transfer is not supported in 1.0.x. Unless you have a brain dead phone, you should be able to use SIP attended transfer in 1.0.x. (that would be the transfer key on the phone) Andrew Nowrot wrote: Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext = 100 parkpos = 1-5 context = parkedcalls parkingtime = 100 transferdigittimeout = 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer = *2 blindxfer = # disconnect = *0 automon = *1 and when I press *2 console says something like this: Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42 (*), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,42) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50 (2), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,50) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Does anyone know what's going on? What should I do to make attended transfer works well? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer - atxfer
Hi, Thank for the Email I'm using 1.0.9 so probably I'm will not have this feature. In which version of Asterisk the DTMF Attended Transfer is supported, in 1.2 Beta? Best wishes Andrew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer - atxfer
Andrew Nowrot wrote: Hi, Thank for the Email I'm using 1.0.9 so probably I'm will not have this feature. In which version of Asterisk the DTMF Attended Transfer is supported, in 1.2 Beta? CVS-HEAD and 1.2Beta1 and later. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call transfer
This is configured on your features.conf file. In there you can see what keys to use to do blind and attended transfers, be sure those lines are not commented out. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Lunes, 01 de Agosto de 2005 01:07 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] call transfer | | | |Hi! | |I have searched answer how can I transfer calls with |asterisk,with no result. |Can you advice me and show some example file how can I use SIP |phone to transfer calls by hitting # and get the Transfer |prompt and enter an extension I want to transfer to? | |Thanks for your answers | | | | |This mail sent through L-secure: http://www.l-secure.net/ | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call transfer
You must use the 't' 'T' options in the Dial() command when placing calls to and from the device. We had extensions that were combinations of SIP and IAX devices and didn't want/need this behavior on all of our devices so we setup our extensions with something as follows: Exten = 1000,1,Dial(Local/IAX-1000/[EMAIL PROTECTED]Local/SIP-1000/[EMAIL PROTECTED], 60, r) [devices] Exten = SIP-1000,1,Dial(SIP-XYZ, 60, tr) Exten = IAX-1000,1,Dial(IAX-ABC, 60, r) That will ring both devices using different dial statements for each. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, August 01, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call transfer Hi! I have searched answer how can I transfer calls with asterisk,with no result. Can you advice me and show some example file how can I use SIP phone to transfer calls by hitting # and get the Transfer prompt and enter an extension I want to transfer to? Thanks for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [SPAM:***** SpamScore] RE: [Asterisk-Users] Call Transfer using SIP clients
On Tuesday 05 July 2005 05:57, Tulika Pradhan wrote:
call transfer works for me fine without any additions in features.conf
by simply using Dial(SIP/${EXTEN},20,tT)
and pressing #number to be transfered to
this works both from caller as well as callee.
tulika
Could you provide me with some more information so I can check where the
differences in our setups are? It would really help to see how you
implemented your extensions and SIP configuration. Could you describe the
following regarding your Asterisk installation:
- Asterisk version
- The SIP clients you use
- Excerpt of extensions.conf, which definitions and contexts do you include
- Excerpt of features.conf, which lines (if any) are in there
- (Maybe) an excerpt of sip.conf, how are the SIP peers configured
I hope you find the time to post these bits and pieces as it will make it
easier for me to debug the situation. I've already tried numerous settings
and combinations of options, but haven't had any luck yet. Thanks in advance
for your precious time.
If anyone else has some ideas regarding my question, feel free to jump in -
the more the merrier.
Sincerely,
Frank
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Re: [SPAM:***** SpamScore] Re: [Asterisk-Users] Call Transfer using SIP clients
On Monday 04 July 2005 16:47, Elwin Andriol wrote: I don't know if this will be of any help to you, but at least I can confirm problems with transfering calls with SIP agents. A little while ago we were having big problems getting transfers using DTMF to work. In that particular situation we were using a mix of only hard SIP devices (BT101's and Elmeg/Snom190's) and ZAP channels. We tried both the stable version of asterisk and the CVS HEAD, but without results (but negative). In the end, we solved the problem by not using DTMF transfers at all, but by using the transfer capabilities of the SIP devices themselves (transfer for and hold buttons). These buttons did not appear to work (correctly) with the stable asterisk version we initially used (1.0.7), but with the CVS HEAD ( 29-MAY-2005) they appear to work just fine. I'm not familiar with soft SIP agents, so I don't know if the ones you use have such build-in transfer capabilities as their hardware counterparts like the BT101's and Snom190's have. I they do, you might wan't to give it a try. This is of course rather a workaround than a solution to your problem. E. Andriol The X-Lite softphone does indeed have a Transfer and Hold in the interface, but the functionality of those buttons appears to have been disabled when the client is connected and registered on the Asterisk server. Pressing the on-screen buttons doesn't have any effect while either having a call or while idle. Related to client-side transferring: I set the canreinvite option in the SIP configuration to no because both clients are behind a NAT / firewall and I read in the documentation that you'd want to disable the canreinvite option in those situations. I haven't had any trouble because of this, as I stated earlier calling and talking is working without hitches. I haven't had the chance to try hardware phones yet, the testing I'm doing at the moment involves softphones only. Now that I think of it, I'll try to setup other applications again which might send DTMFs in a different form compared to X-Lite. In the meantime thanks in advance to everyone involved in this thread now and in the future. Sincerely, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer using SIP clients
Frank Schoep wrote: The X-Lite softphone does indeed have a Transfer and Hold in the interface, but the functionality of those buttons appears to have been disabled when the client is connected and registered on the Asterisk server. Pressing the on-screen buttons doesn't have any effect while either having a call or while idle. I'm pretty sure it's disabled on purpose on the free Linux version of the phone. I remember reading that somewhere on their site once upon a time. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer using SIP clients
Frank Schoep wrote: Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently using Asterisk CVS as of July 4th 2005. The only means of communication at the moment is the XTen X-Lite SIP Client, I already added the following entries to my sip.conf configuration file: [frank] canreinvite=no type=friend secret=frank username=frank nat=yes host=dynamic [test] canreinvite=no type=friend secret=test username=test nat=yes host=dynamic The SIP setup is working without a problem, the X-Lite application correctly registers the users and I can set up calls between them. I've also tested queues and they work without a problem, too. Next up is my extensions configuration, of which the interesting section now looks like this: [default] include = general ; longshot, added out of desparation include = parkedcalls ; longshot, added out of desparation include = featuremap ; longshot, added out of desparation exten = 800,1,Answer exten = 800,2,Dial(SIP/frank,20,tT) exten = 800,3 Hangup exten = 802,1,Answer exten = 802,2,Dial(SIP/test,20,tT) exten = 802,3 Hangup Notice the inclusion of several contexts that should or would have to be defined in the features configuration. My features.conf looks something like this, I trimmed the 'general' section for brevity: [general] ; (trimmed) default options [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer My testing scenario starts as follows: - log in both X-Lite SIP clients - from the 'test' phone, call extension 800 - on X-Lite client 'frank' accept the call - talk to eachother At this point I want to transfer to call to another extension, also defined in sip.conf but unlisted here. The problem is that nothing happens when I press the #1 or *2 keys in the 'frank' X-Lite client. I also tested these key combinations on the 'test' X-Lite client during the call, but that also had not effect. I searched the web and the mailing list archive for a solution, and if I recall correctly, someone stated that call transfer is only available for calls originating from the PSTN. Is this correct, also in regard of the current version of Asterisk? Has anyone got an idea how to get call transfer to work? One thing I tried was to change the DTMF settings in the clients, so they are sent in-band, but this also didn't help. Should I revert this option? Thanks in advance for your time and patience. Sincerely, Frank Schoep ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I don't know if this will be of any help to you, but at least I can confirm problems with transfering calls with SIP agents. A little while ago we were having big problems getting transfers using DTMF to work. In that particular situation we were using a mix of only hard SIP devices (BT101's and Elmeg/Snom190's) and ZAP channels. We tried both the stable version of asterisk and the CVS HEAD, but without results (but negative). In the end, we solved the problem by not using DTMF transfers at all, but by using the transfer capabilities of the SIP devices themselves (transfer for and hold buttons). These buttons did not appear to work (correctly) with the stable asterisk version we initially used (1.0.7), but with the CVS HEAD ( 29-MAY-2005) they appear to work just fine. I'm not familiar with soft SIP agents, so I don't know if the ones you use have such build-in transfer capabilities as their hardware counterparts like the BT101's and Snom190's have. I they do, you might wan't to give it a try. This is of course rather a workaround than a solution to your problem. E. Andriol -- --- HeuvelTop ICT Diensten v.o.f. --- There are management solutions to technical problems, but no technical solutions to management problems --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Transfer using SIP clients
call transfer works for me fine without any additions in features.conf
by simply using Dial(SIP/${EXTEN},20,tT)
and pressing #number to be transfered to
this works both from caller as well as callee.
tulika
From: Frank Schoep [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Call Transfer using SIP clients
Date: Mon, 4 Jul 2005 16:11:13 +0200
Hello all,
First of all, let me apologize about the length of this message, but I
suppose
it was necessary to include the details.
I've spent quite some time already trying to get the call transfer function
to
work on my Asterisk installation. Let me first describe the general
situation
of the setup I am using, so you might be able to pinpoint the cause of the
problem.
I'm currently using Asterisk CVS as of July 4th 2005. The only means of
communication at the moment is the XTen X-Lite SIP Client, I already added
the following entries to my sip.conf configuration file:
[frank]
canreinvite=no
type=friend
secret=frank
username=frank
nat=yes
host=dynamic
[test]
canreinvite=no
type=friend
secret=test
username=test
nat=yes
host=dynamic
The SIP setup is working without a problem, the X-Lite application
correctly
registers the users and I can set up calls between them. I've also tested
queues and they work without a problem, too. Next up is my extensions
configuration, of which the interesting section now looks like this:
[default]
include = general ; longshot, added out of desparation
include = parkedcalls ; longshot, added out of desparation
include = featuremap ; longshot, added out of desparation
exten = 800,1,Answer
exten = 800,2,Dial(SIP/frank,20,tT)
exten = 800,3 Hangup
exten = 802,1,Answer
exten = 802,2,Dial(SIP/test,20,tT)
exten = 802,3 Hangup
Notice the inclusion of several contexts that should or would have to be
defined in the features configuration. My features.conf looks something
like
this, I trimmed the 'general' section for brevity:
[general]
; (trimmed) default options
[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0 ; Disconnect
automon = *1 ; One Touch Record
atxfer = *2 ; Attended transfer
My testing scenario starts as follows:
- log in both X-Lite SIP clients
- from the 'test' phone, call extension 800
- on X-Lite client 'frank' accept the call
- talk to eachother
At this point I want to transfer to call to another extension, also defined
in
sip.conf but unlisted here. The problem is that nothing happens when I
press the #1 or *2 keys in the 'frank' X-Lite client. I also tested
these
key combinations on the 'test' X-Lite client during the call, but that also
had not effect.
I searched the web and the mailing list archive for a solution, and if I
recall correctly, someone stated that call transfer is only available for
calls originating from the PSTN. Is this correct, also in regard of the
current version of Asterisk? Has anyone got an idea how to get call
transfer
to work?
One thing I tried was to change the DTMF settings in the clients, so they
are
sent in-band, but this also didn't help. Should I revert this option?
Thanks in advance for your time and patience.
Sincerely,
Frank Schoep
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Re: [Asterisk-Users] Call Transfer Problem
Adam Robins wrote: The double-star now works great. If I press it while on a call, I go into transfer mode. The problem is that the # still works as well! Shouldn't the atzfer specification turn off the #? Blind transfers are on '#' by default, so you may need to move them to another sequence as well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
Henry Devito wrote: I just bought one of these zyxel wireless phones, of course there is no transfer key. Is there a patch for the stable 1.0.7 that I can implement # or any other key or combination to initiate a transfer? I looked briefly through the wiki and archived lists and didn't see much. show application dial Pay special attention to the t/T options. Those options are for devices that are too stupid or brain dead to have a transfer key that works. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
Thanks, Don't know how I could have missed that, This works on incoming calls to the station and calls from station to station. How do I make it work if I dial out over a zap channel and then want to transfer to another extension the # doesn't do anything except generate tone on the line. N0 transfer, - Original Message - From: Eric Wieling aka ManxPower [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 28, 2005 12:26 PM Subject: Re: [Asterisk-Users] Call transfer Henry Devito wrote: I just bought one of these zyxel wireless phones, of course there is no transfer key. Is there a patch for the stable 1.0.7 that I can implement # or any other key or combination to initiate a transfer? I looked briefly through the wiki and archived lists and didn't see much. show application dial Pay special attention to the t/T options. Those options are for devices that are too stupid or brain dead to have a transfer key that works. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer Features
Use ## it's in features.conf On Tue, 22 Mar 2005 07:42:27 -0700, Damon Estep [EMAIL PROTECTED] wrote: Looking for a liitle help if anyone has dealt with this; The options on dial and queue of t (allow called party to transfer call) and T (allow calling aprty to transfer call) seem to work fine (as long as you do not confuse them with the same t and T that indicate timeout!). The problem I am having is the use of the # key to do so. Many times a caller will palce a call to an IVR that requires the use of the # key to access a feature on the remote IVR, but * intercepts the # and offers the transfer prompt (as expected). The solution seems to be to change the feature to use a different key, although I do not know how. I am sure I could find it on the wiki, but it seems to be down this morning? Any suggestinos on a transfer key sequence that does not interfere with external IVRs that often? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
On Tue, 8 Mar 2005 14:17:23 -0300 Alejandro G [EMAIL PROTECTED] wrote: I have 2 asterisk box in different locations. When I received a call in one location and want to transfer it to an extension in the other location the external call is hanged up when the person who is transfering the call hangs up. The sequence is like this: 1. Call is received and attended by person 1 in extension 3000 in location 1 2. Person 1 press flash and dials extension 4000 in location 2 3. Person 2 in extension 4000 i location 2 pick up the call and talk to person 1 4. Person 1 hangs up and the external call is hanged up. Is anything wrong? Thanks. Alejandro Ghergherian Post your conf file sections that are relavant from extensions, sip or iax That way we can see how you are handling things and be able to see what might be wrong. Also, bring up a cli and monitor what is happening during the transfer. Then copy and paste that to the list. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer questions
At 05:44 AM 3/7/2005, you wrote: Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer also can the caller hear MOH while I am talking to person I am transfering the call to what would I need to do this just point me in the right direction and i'll go read some more... I using so far is T in dial() Thanks sorry for the noob question also tried the following without luck [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 it still seems to want to accept only # as transfer I am running Asterisk CVS-v1-0-03/07/05-06:50:06 Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer questions
also tried the following without luck [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 it still seems to want to accept only # as transfer I am running Asterisk CVS-v1-0-03/07/05-06:50:06 You are running V1.0.x stable of asterisk. Tthe attended transfer feature is only available in CVS-HEAD, which at some point (June ?) will become 1.1.x stable ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call transfer to conference call
You need to send a Manager command(Redirect Action) to the asterisk server. (BYou can do this by connecting to the manager API through any kind of (Btelnet-type connector in any number of programming languages: Perl, C, PHP, (Betc.. (B (BTake a look at the WIKI for more info: (B (Bhttp://www.voip-info.org/wiki-Asterisk+manager+API (B (BThe redirect of the existing conversation to the conference room(meetme) (Bwould look something like this: (B (BAction: Redirect (BChannel: Zap/73-1 (BExtraChannel: SIP/199testphone-1f3c (BExten: 8600029 (BContext: default (BPriority: 1 (B (Bwhere 8600029 is the meetme room. (B (BHope that helps, (B (BMATT--- (B (B (B-Original Message- (BFrom: Kuniyoshi Murata [mailto:[EMAIL PROTECTED] (BSent: Monday, January 03, 2005 9:26 PM (BTo: Asterisk Users Mailing List - Non-Commercial Discussion (BSubject: [Asterisk-Users] call transfer to conference call (B (B (BHi, (B (BI have following setup already. (B (BPSTN call via zap channel is working, Xlite via sip channel is working, and (Bconference call is working. (B (BAnd here is what I want to do. (B (BA. My friends are making conference call in a conference room and all (Bclients are Xlite. (BB. I make a call from my Xlite to PSTN phone of the guest speaker I want to (Binvite to the conference call (BC. I will transfer my call with the guest to the conference call that is (Balready taking place (that is A) (B (BOf course I can do A and B, but to do C what kind of command, operation, (Band/or script are needed? (B (BAny comments, directions and references are welcome. (B (BThanks in advance (BKuni (B (B-- (BKuniyoshi Murata.iChat/AIM:macwebcaster (BEnglish-Japanese Interpreter mailto:[EMAIL PROTECTED] (BMacintosh Webcast Specialisthttp://www.macwebcaster.com (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer Problem
On Fri, 8 Oct 2004, Michael Nolan wrote: Hi ! I have checked my asterisk. It contains this patch or thBis patch is already compiled into it. can you plz guide me as to how i can make use of it ? I have pressed '#' but it doesnot give me any dial tone. Are there any special changes that need to be done in extensions.conf to make it work ? plz help me in this regard. Usman. This patch works a treat for us: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Makes all # transfers attended, but the transfer button on the phones can still be used for blind transfers from our SIP phones. Cheers, Michael On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users, I am having a prblem using attended call transfer with asterisk. Actually my sip phone does not seem to support it. Can i use attended call transfer using the dial plan ... ??? means can somehow messing up with extesnions.conf I can get attended call transfer ? And yes also is there any way I can do it with AGI scripting ? Any AGI similar examples will be a lot of help. Thanks ! Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer Problem
you need the x or X option to your Dial command. show application dial is your friend ... cheers Michael On Mon, 11 Oct 2004 08:37:36 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Fri, 8 Oct 2004, Michael Nolan wrote: Hi ! I have checked my asterisk. It contains this patch or thBis patch is already compiled into it. can you plz guide me as to how i can make use of it ? I have pressed '#' but it doesnot give me any dial tone. Are there any special changes that need to be done in extensions.conf to make it work ? plz help me in this regard. Usman. This patch works a treat for us: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Makes all # transfers attended, but the transfer button on the phones can still be used for blind transfers from our SIP phones. Cheers, Michael On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users, I am having a prblem using attended call transfer with asterisk. Actually my sip phone does not seem to support it. Can i use attended call transfer using the dial plan ... ??? means can somehow messing up with extesnions.conf I can get attended call transfer ? And yes also is there any way I can do it with AGI scripting ? Any AGI similar examples will be a lot of help. Thanks ! Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone
Title: Mensaje Push send after you number, srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de James DuttonEnviado el: jueves, 05 de agosto de 2004 12:28Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone I have two Grandstream Budgetone 100 phones connected to my local asterisk server. I am able to receive incoming calls, and place outgoing calls, but have two problems... 1) I cannot transfer calls between the two phones. Pressing transfer takes me to a dial tone, I key in the internal number then press # or transfer, and the original call is cut off and the other internal phone does not ring. 2) I cannot hear an outgoing ringing tone when placing the call. I would be extremely grateful to anyone out who has experience of these phones and can help. Regards James Dutton
Re: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone
- Original Message - From: James Dutton To: [EMAIL PROTECTED] Sent: Thursday, August 05, 2004 6:27 AM Subject: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone I have two Grandstream Budgetone 100 phones connected to my local asterisk server. I am able to receive incoming calls, and place outgoing calls, but have two problems... 1) I cannot transfer calls between the two phones. Pressing transfer takes me to a dial tone, I key in the internal number then press # or transfer, and the original call is cut off and the other internal phone does not ring. try pressing send instead of # or transfer 2) I cannot hear an outgoing ringing tone when placing the call. add r to your dial statement in extensions.conf I would be extremely grateful to anyone out who has experience of these phones and can help. Regards James Dutton
Re: [Asterisk-Users] call transfer with consultation
For example, when an input call comes through X100P, my Zap/3 extension rings. I pickup Zap/3 and I want to transfer the call to Zap/4, but before to establish the call between X100P and Zap/4 I need to request Zap/4 for answering the call. Currently not possible, although here is a workaround since you are using Zap interfaces: Call comes in and you answer it. Hook flash (briefly hang up and pick up the phone again) -- caller is on hold and you can dial the extension you want to transfer it to. Talk to the extension Hook flash again, and now you, the extension and the caller are in a 3-way call. Hang up -- the call is now transfered. Hope this helps. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call transfer
WARNING[1242952640]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice This is what I get And a crash Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 17 November 2003 5:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call transfer On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote: Does anyone know how to make Calls auto transfer to a mobile if no one answers ?? suppose your mobile number is +923008508070 exten = 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX exten = 15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
Hi Mick, It's going to be hard for anybody here on the list to help you, unless you are more specific, ie, what you did exactly to get a crash, and console output (with verbose set) debugs, logs (under /var/log/asterisk) and some configuration files. We'll be in a better position to help you then without trying to be mind readers. Paul - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 6:49 PM Subject: RE: [Asterisk-Users] Call transfer WARNING[1242952640]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice This is what I get And a crash Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 17 November 2003 5:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call transfer On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote: Does anyone know how to make Calls auto transfer to a mobile if no one answers ?? suppose your mobile number is +923008508070 exten = 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX exten = 15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote: Does anyone know how to make Calls auto transfer to a mobile if no one answers ?? suppose your mobile number is +923008508070 exten = 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX exten = 15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
CW_ASN wrote: I use 3Party using flash key and dialing the extension. When the other ATA answer the call, I press flash again. I test Call Transfer using # key (#ext#). If you know another way to do that, please let me know. I'm tearing my hair out trying to exercise a variation on this theme. I'm mad from trying, so there may be some realy easy thing that is escaping me here. What I want to do is answer a call, put the caller on hold, dial up another extension and speak briefly with the person who answers (e.g. I have Mr. Faltzernaust on the line) and then bow out and leave the caller and the callee to talk. When I try the 3party the whole thing goes to s**t when I hang up, and in a transfer I don't get that opportunity to announce the caller. Is there an easy way to do what I want to do here? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
Hi Gus, I have all those problems too, but all gone when update to the latest Asterisk CVS. Now I can use unattended transfer on ATA with '#' or Flash. Check the following settings in ATA (I presume that SIP is used): CallFeatures: 0x0ff80ff8 ConnectMode:0x00460400 Tell me exactly how have you tried to do the transfer (step by step). Best regards, Dan - Original Message - From: ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 19, 2003 12:51 AM Subject: [Asterisk-Users] Call transfer ATA186 Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
Hi, You can find the standard procedure to do this on ATA here: http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html#1015891 for unattended transfer and: http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html#1015900 for attended transfer. They both work for me now. You can use '#' procedure too. BR, Dan - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 19, 2003 9:42 AM Subject: Re: [Asterisk-Users] Call transfer ATA186 CW_ASN wrote: I use 3Party using flash key and dialing the extension. When the other ATA answer the call, I press flash again. I test Call Transfer using # key (#ext#). If you know another way to do that, please let me know. I'm tearing my hair out trying to exercise a variation on this theme. I'm mad from trying, so there may be some realy easy thing that is escaping me here. What I want to do is answer a call, put the caller on hold, dial up another extension and speak briefly with the person who answers (e.g. I have Mr. Faltzernaust on the line) and then bow out and leave the caller and the callee to talk. When I try the 3party the whole thing goes to s**t when I hang up, and in a transfer I don't get that opportunity to announce the caller. Is there an easy way to do what I want to do here? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
At 1:42 AM -0500 8/19/03, Brian Capouch wrote: From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call transfer ATA186 Reply-To: [EMAIL PROTECTED] Date: Tue, 19 Aug 2003 01:42:53 -0500 CW_ASN wrote: I use 3Party using flash key and dialing the extension. When the other ATA answer the call, I press flash again. I test Call Transfer using # key (#ext#). If you know another way to do that, please let me know. I'm tearing my hair out trying to exercise a variation on this theme. I'm mad from trying, so there may be some realy easy thing that is escaping me here. What I want to do is answer a call, put the caller on hold, dial up another extension and speak briefly with the person who answers (e.g. I have Mr. Faltzernaust on the line) and then bow out and leave the caller and the callee to talk. When I try the 3party the whole thing goes to s**t when I hang up, and in a transfer I don't get that opportunity to announce the caller. Is there an easy way to do what I want to do here? Thx. B. If it's any solace to you, there is no way I know of that one can do supervised call transfer (what you describe above) on an ATA-186. The # key trick (appending a t on your Dial statements) allows you to transfer, but it's an unsupervised transfer. That trickery is done 100% in Asterisk, so you may be able to hack the source code to get it working with some different technique. The workaround is: - call party #1, establish call - hit flash, dial new number, hit # (the # is locally interpreted by the ATA as finished dialing character) - talk to the third party; tell them a call is about to come in. - have the third party hang up - hit flash again, bringing you back to party #1 - hit # and type in the number of third party, type # (this time, interpreted by Asterisk as finished dialing character) - hang up Sucks, doesn't it? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
Hi John, - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 19, 2003 10:07 AM Subject: Re: [Asterisk-Users] Call transfer ATA186 If it's any solace to you, there is no way I know of that one can do supervised call transfer (what you describe above) on an ATA-186. The # key trick (appending a t on your Dial statements) allows you to transfer, but it's an unsupervised transfer. That trickery is done 100% in Asterisk, so you may be able to hack the source code to get it working with some different technique. It works for me now using Flash key. When updated to the latest CVS and setting the call parameters in ATA config the unattended transfer works on ATA too now. The workaround is: - call party #1, establish call - hit flash, dial new number, hit # (the # is locally interpreted by the ATA as finished dialing character) - talk to the third party; tell them a call is about to come in. - have the third party hang up - hit flash again, bringing you back to party #1 - hit # and type in the number of third party, type # (this time, interpreted by Asterisk as finished dialing character) - hang up Sucks, doesn't it? For me the workaround was to define a dial loop in extensions.conf when a phone is busy (call waiting must be activated). Then do like that. - answer the call on ATA - pres Flash key - call the final destination - wait to answer and inform it about the call - close the phone - fastbusy on final destination - close the final destination - final destination is automatically called by Asterisk from the initial caller. - establish the call between them. All you need is to keep dialing the final destination (in a loop) if busy, till it rings. It seems that with the latest CVS update and the correct parameters in ATA config page this works without the need to do this trick. Just do like that: - answer the call on ATA - pres Flash key - call the final destination - wait to answer and inform it about the call - close the phone - MOH is stopped and the call is automatically establish between them . Tested several times with different configurations: 7960-ATA-ATA X-Lite-7960-ATA 7960-X-Lite-ATA ATA-7960-ATA ATA-ATA-7960 and it works for me now. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
At 12:07 AM -0700 8/19/03, John Todd wrote: At 1:42 AM -0500 8/19/03, Brian Capouch wrote: CW_ASN wrote: I use 3Party using flash key and dialing the extension. When the other ATA answer the call, I press flash again. I test Call Transfer using # key (#ext#). If you know another way to do that, please let me know. I'm tearing my hair out trying to exercise a variation on this theme. I'm mad from trying, so there may be some realy easy thing that is escaping me here. What I want to do is answer a call, put the caller on hold, dial up another extension and speak briefly with the person who answers (e.g. I have Mr. Faltzernaust on the line) and then bow out and leave the caller and the callee to talk. When I try the 3party the whole thing goes to s**t when I hang up, and in a transfer I don't get that opportunity to announce the caller. Is there an easy way to do what I want to do here? Thx. B. If it's any solace to you, there is no way I know of that one can do supervised call transfer (what you describe above) on an ATA-186. The # key trick (appending a t on your Dial statements) allows you to transfer, but it's an unsupervised transfer. That trickery is done 100% in Asterisk, so you may be able to hack the source code to get it working with some different technique. The workaround is: - call party #1, establish call - hit flash, dial new number, hit # (the # is locally interpreted by the ATA as finished dialing character) - talk to the third party; tell them a call is about to come in. - have the third party hang up - hit flash again, bringing you back to party #1 - hit # and type in the number of third party, type # (this time, interpreted by Asterisk as finished dialing character) - hang up Sucks, doesn't it? JT Duh. I shouldn't work so late. All this time I had simply been forgetting to hang up on people. :) I suppose this makes sense; it's half of a three-way call, except after you talk to the third party, you just hang up. The ATA then redials the third party and connects the two together. The RTP stream no longer goes through the ATA after the redial, if you're curious. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html#1015900 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
It might be possible to implement BYE also style transfer which is totally deprecated in SIP but appears to be (at least as of a few months ago) what the ATA's used. If you want to add a bug to the bug tracker (including SIP debug) from your attmpted call, I can take a look at it. Mark On Tue, 19 Aug 2003, Brian Capouch wrote: CW_ASN wrote: I use 3Party using flash key and dialing the extension. When the other ATA answer the call, I press flash again. I test Call Transfer using # key (#ext#). If you know another way to do that, please let me know. I'm tearing my hair out trying to exercise a variation on this theme. I'm mad from trying, so there may be some realy easy thing that is escaping me here. What I want to do is answer a call, put the caller on hold, dial up another extension and speak briefly with the person who answers (e.g. I have Mr. Faltzernaust on the line) and then bow out and leave the caller and the callee to talk. When I try the 3party the whole thing goes to s**t when I hang up, and in a transfer I don't get that opportunity to announce the caller. Is there an easy way to do what I want to do here? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
Works for me.. I can press # and dial the ext and press # to transfer a call. www.bkw.org/~brian/ata.html for the settings I used in my ATA bkw On Mon, 18 Aug 2003, ASN wrote: Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
How exactly does you 3Party calling work? ;) Fred ASN wrote: Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
I use 3Party using flash key and dialing the extension. When the other ATA answer the call, I press flash again. I test Call Transfer using # key (#ext#). If you know another way to do that, please let me know. Regards, Gus - Original Message - From: Fredrik Hedberg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 7:15 PM Subject: Re: [Asterisk-Users] Call transfer ATA186 How exactly does you 3Party calling work? ;) Fred ASN wrote: Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
I will copy the configurations and let you know (may be some parameter is wrong). Thanks a lot! Gus - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 7:03 PM Subject: Re: [Asterisk-Users] Call transfer ATA186 Works for me.. I can press # and dial the ext and press # to transfer a call. www.bkw.org/~brian/ata.html for the settings I used in my ATA bkw On Mon, 18 Aug 2003, ASN wrote: Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Excellent idea mate, Now I am able to do what I wanted with Great help from Jeremy McNamara. Thanks alot Foong - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hi Foong, But then... who and when will trigger the transfer between the two remote extensions? I think to something like that. One of the extension calls a special number, entering a password (or check after the Caller ID). Asterisk close the call, wait for answer Call the second extension, wait for answer Then, in some way (eventually through a conference mode using local CONSOLE as master) bridge the two calls. What do you think about that? Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:30 AM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Dan, Asterisk is suppose to trigger the transfer when it successfully call both extensions Do you mean I have to create conference room for every call? that would not be practicle. Or do you have a example dialplan to to illustrate you suggestion? Actually, we have a client that is too lazy to do all the dialing, he want a system that will call him and also the person he wanted to call, just like some receptionists do theese days. The different is that asterisk is taking over the receptionist's job thanks Foong Hi Foong, But then... who and when will trigger the transfer between the two remote extensions? I think to something like that. One of the extension calls a special number, entering a password (or check after the Caller ID). Asterisk close the call, wait for answer Call the second extension, wait for answer Then, in some way (eventually through a conference mode using local CONSOLE as master) bridge the two calls. What do you think about that? Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:30 AM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer, Budgettone 100
park the call On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote: hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and then the new extension .. cheers Dave --- Email sent using AnyEmail (http://netbula.com/anyemail/) Netbula LLC is not responsible for the content of this email ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer, Budgettone 100
Last I checked, SIP transfer to park doesn't work .. only way to do it is using T and a # transfer .. which is ugly. Has this been fixed? -d At 10:51 AM 7/30/2003 +0200, you wrote: park the call On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote: hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and then the new extension .. cheers Dave --- Email sent using AnyEmail (http://netbula.com/anyemail/) Netbula LLC is not responsible for the content of this email ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Yes, I second to that idea. I think thats only available option to put them in a local conference. Rgds Manoj K Gupta - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 2:04 PM Subject: Re: [Asterisk-Users] Call Transfer Hi Foong, But then... who and when will trigger the transfer between the two remote extensions? I think to something like that. One of the extension calls a special number, entering a password (or check after the Caller ID). Asterisk close the call, wait for answer Call the second extension, wait for answer Then, in some way (eventually through a conference mode using local CONSOLE as master) bridge the two calls. What do you think about that? Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:30 AM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hi I would like to further ask if it is possible to transfer a call from openphone to pstn. i.e. i use openphone and asterisk -oh323 channel driver to make a call to a PSTN number through zap channel connected on that end.Then i wanna transfer that PSTN number to some other openphone extension/alias May i have a look at your extension to conf, as i am not clear with how to implement this. Rgds Manoj k Gupta - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 2:00 PM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hello But If i do that I have to create lots of conference room if I have lots of caller. Foong - Original Message - From: Sip Rtp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 5:44 PM Subject: Re: [Asterisk-Users] Call Transfer Yes, I second to that idea. I think thats only available option to put them in a local conference. Rgds Manoj K Gupta - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 2:04 PM Subject: Re: [Asterisk-Users] Call Transfer Hi Foong, But then... who and when will trigger the transfer between the two remote extensions? I think to something like that. One of the extension calls a special number, entering a password (or check after the Caller ID). Asterisk close the call, wait for answer Call the second extension, wait for answer Then, in some way (eventually through a conference mode using local CONSOLE as master) bridge the two calls. What do you think about that? Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:30 AM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hi Sip,
I achieve that by adding the following extension into extension.conf:
exten = _9,1,Dial(H323/{EXTEN:1})
foong
- Original Message -
From: Sip Rtp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 5:45 PM
Subject: Re: [Asterisk-Users] Call Transfer
Hi
I would like to further ask if it is possible to
transfer a call from
openphone to pstn. i.e. i use openphone and asterisk
-oh323 channel driver
to make a call to a PSTN number through zap channel
connected on that
end.Then i wanna transfer that PSTN number to some
other openphone
extension/alias
May i have a look at your extension to conf, as i am
not clear with how to
implement this.
Rgds
Manoj k Gupta
- Original Message -
From: Chee Foong [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 2:00 PM
Subject: Re: [Asterisk-Users] Call Transfer
Hello Dan,
Thanks for you reply.
Base on you recomendation using the 'T' argument. I
manage to do call
transfer an it works really well.
My problem comes when my boss comes out with a
superb idea where the
transfering process is automated without involving a
human :(
Say asterisk get 2 numbers (from database, text
file, etc), one belongs
party A and the other belongs to party B. Asterisk
will calls both parties
and do the tranfer automatically. In another words,
asterisk is resposible
to 'press' the '#' to do the transfer. I don't this
can be achieve in the
extension.conf not matter how you structure you dial
plan.
Perhaps, the only way is to write a apps and plug it
into asterisk like
all
the asterisk modules such as Meetme.
Any ideas?
Foong
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 3:42 PM
Subject: Re: [Asterisk-Users] Call Transfer
Hi,
It works if you put the 'T' switch in the dial
line.
You can then transfer the call from the caller.
I have tested it in the folllowing configuration
and it works:
Call from a Cisco 7960 to an ATA 186.
Select 'Transfer on 7960
Call another extension (X-Lite)
Select again transfer on 7960.
The call remain between ATA and X-Lite.
This is what you need?
BR,
Dan
- Original Message -
From: Chee Foong [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 7:08 AM
Subject: [Asterisk-Users] Call Transfer
Hello all,
I am in a situation where I need to use asterisk
to call someone say
Party
A. After the call to Party A got through, asterisk
will put Party A on
hold,
then asterisk will call Party B. If call to Party
B got through,
asterisk
will transfer Party A to Party B.
I wonder if this features is implemented into
asterisk. I have found a
post
in asterisk mailing list:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
but that doesn't help much.
If this features is not implemented, can anyone
give me some point on
how
to
implement this in asterisk? Do I need to write an
app like the Dial apps
for
asterisk to load at start up?
thanks
Foong
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Re: [Asterisk-Users] Call Transfer
Foong, Actually, we have a client that is too lazy to do all the dialing, he want a system that will call him and also the person he wanted to call, just like some receptionists do theese days. The different is that asterisk is taking over the receptionist's job ... then... who decide when the call must be initiated and how? Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
There is no need to create a Meeting Room... just to initiate a conference in three... - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 1:02 PM Subject: Re: [Asterisk-Users] Call Transfer Hello But If i do that I have to create lots of conference room if I have lots of caller. Foong - Original Message - From: Sip Rtp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 5:44 PM Subject: Re: [Asterisk-Users] Call Transfer Yes, I second to that idea. I think thats only available option to put them in a local conference. Rgds Manoj K Gupta - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 2:04 PM Subject: Re: [Asterisk-Users] Call Transfer Hi Foong, But then... who and when will trigger the transfer between the two remote extensions? I think to something like that. One of the extension calls a special number, entering a password (or check after the Caller ID). Asterisk close the call, wait for answer Call the second extension, wait for answer Then, in some way (eventually through a conference mode using local CONSOLE as master) bridge the two calls. What do you think about that? Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:30 AM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users
Re: [Asterisk-Users] Call Transfer
Thanks Andy Will try that Thanks again. Foong - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
