hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping when it comes to call transfering ... exten => s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds ... can anybody gv me an idea? Thank you inadvance, Surajee --------------This mail sent through OmniBIS.com--------------
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030729/aacfdf45/attachment.htm
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" <dtoma@fx.ro> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 2:04 PM Subject: Re: [Asterisk-Users] Call Transfer> Hi Foong, > > But then... who and when will trigger the transferbetween 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 conferencemode using local CONSOLE> as master) bridge the two calls. > What do you think about that? > > Dan > > > ----- Original Message ----- > From: "Chee Foong" <cheefoong@inovas.com> > To: <asterisk-users@lists.digium.com> > 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 asuperb idea where the> > transfering process is automated without involvinga human :(> > > > Say asterisk get 2 numbers (from database, textfile, etc), one belongs> > party A and the other belongs to party B. Asteriskwill calls both parties> > and do the tranfer automatically. In anotherwords, asterisk is resposible> > to 'press' the '#' to do the transfer. I don'tthis can be achieve in the> > extension.conf not matter how you structure youdial plan.> > > > Perhaps, the only way is to write a apps and plugit into asterisk like> all > > the asterisk modules such as Meetme. > > > > Any ideas? > > > > > > Foong > > > > ----- Original Message ----- > > From: "Dan" <dtoma@fx.ro> > > To: <asterisk-users@lists.digium.com> > > 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 dialline.> > > > > > You can then transfer the call from the caller. > > > I have tested it in the folllowing configurationand 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" <cheefoong@inovas.com> > > > To: <asterisk-users@lists.digium.com> > > > 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 asteriskto 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 toParty B got through,> asterisk > > > will transfer Party A to Party B. > > > > > > I wonder if this features is implemented intoasterisk. 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 anyonegive me some point on> how > > to > > > implement this in asterisk? Do I need to writean app like the Dial apps> > for > > > asterisk to load at start up? > > > > > > > > > thanks > > > > > > Foong > > > > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > >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> > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >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
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" <cheefoong@inovas.com> To: <asterisk-users@lists.digium.com> 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. Imanage to do call> transfer an it works really well. > > My problem comes when my boss comes out with asuperb idea where the> transfering process is automated without involving ahuman :(> > Say asterisk get 2 numbers (from database, textfile, etc), one belongs> party A and the other belongs to party B. Asteriskwill calls both parties> and do the tranfer automatically. In another words,asterisk is resposible> to 'press' the '#' to do the transfer. I don't thiscan be achieve in the> extension.conf not matter how you structure you dialplan.> > Perhaps, the only way is to write a apps and plug itinto asterisk like all> the asterisk modules such as Meetme. > > Any ideas? > > > Foong > > ----- Original Message ----- > From: "Dan" <dtoma@fx.ro> > To: <asterisk-users@lists.digium.com> > 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 dialline.> > > > You can then transfer the call from the caller. > > I have tested it in the folllowing configurationand 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" <cheefoong@inovas.com> > > To: <asterisk-users@lists.digium.com> > > 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 asteriskto call someone say Party> > A. After the call to Party A got through, asteriskwill put Party A on> hold, > > then asterisk will call Party B. If call to PartyB got through, asterisk> > will transfer Party A to Party B. > > > > I wonder if this features is implemented intoasterisk. 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 anyonegive me some point on how> to > > implement this in asterisk? Do I need to write anapp like the Dial apps> for > > asterisk to load at start up? > > > > > > thanks > > > > Foong > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > >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 __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com
Does anyone know how to make Calls auto transfer to a mobile if no one answers ?? Regards Mick West
On Mon, 17 Nov 2003 mick@netexpress.com.au 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
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: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of wasim@convergence.com.pk Sent: Monday, 17 November 2003 5:14 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Call transfer On Mon, 17 Nov 2003 mick@netexpress.com.au 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Dear All, How can I make a call transfer and line release after connected? I've found the Transfer(Zap/..) is not working as expect.... Thanks for your help! regards, R Wong
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
On Tue, 8 Mar 2005 14:17:23 -0300 "Alejandro G" <ag10@fibertel.com.ar> 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
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.
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/
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/
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: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |laine.marko@porilainen.com |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 | |
Hello, I have my *@HOME 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
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 Asterisk@Home 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? -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000
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 David.Morrow@Autodata.Net 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 Administrator@autodata.net <mailto:Administrator@autodata.net> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060113/a4085af7/attachment.htm