[asterisk-users] Transfering Calls back on the same PRI
Here is my hardware configuration TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk The PBX is a Siemens Hicom 200 EX (Model 80) We are connecting between the PBX and Asterisk using QSIG switch type. What I want to do is the following: 1. Call comes from TELCO via PRI1 and enters PBX 2. PBX Routes call to Asterisk via PRI2 3. Asterisk does some call handling (IVR) 4. Call needs to be transfered to an extension on the PBX. I can easily set up a dial command to pass the call back to the PBX from Asterisk along PRI2 but this uses 2 B Channels. How do I tell asterisk to send a transfer request to the PBX so Asterisk is out of the loop? Thanks, Ron ___ -- 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
[asterisk-users] Transfering Calls back on the same PRI
Here is my hardware configuration TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk The PBX is a Siemens Hicom 200 EX (Model 80) We are connecting between the PBX and Asterisk using QSIG switch type. What I want to do is the following: 1. Call comes from TELCO via PRI1 and enters PBX 2. PBX Routes call to Asterisk via PRI2 3. Asterisk does some call handling (IVR) 4. Call needs to be transfered to an extension on the PBX. I can easily set up a dial command to pass the call back to the PBX from Asterisk along PRI2 but this uses 2 B Channels. How do I tell asterisk to send a transfer request to the PBX so Asterisk is out of the loop? Thanks, Ron -- Ron Joffe Siena Tech, Inc. 3319 Willow Glen Drive Oak Hill, VA 20171 (919) 928-0404 ___ -- 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] Transfering Calls back on the same PRI
On Fri, Oct 17, 2008 at 5:24 PM, Ron Joffe [EMAIL PROTECTED] wrote: Here is my hardware configuration TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk The PBX is a Siemens Hicom 200 EX (Model 80) We are connecting between the PBX and Asterisk using QSIG switch type. What I want to do is the following: 1. Call comes from TELCO via PRI1 and enters PBX 2. PBX Routes call to Asterisk via PRI2 3. Asterisk does some call handling (IVR) 4. Call needs to be transfered to an extension on the PBX. I can easily set up a dial command to pass the call back to the PBX from Asterisk along PRI2 but this uses 2 B Channels. How do I tell asterisk to send a transfer request to the PBX so Asterisk is out of the loop? Thanks, Ron -- Ron Joffe Siena Tech, Inc. 3319 Willow Glen Drive Oak Hill, VA 20171 (919) 928-0404 I would engineer the system so that Asterisk is in the middle rather than the far end. Is there a reason why you don't want to or cannot do that? TELCO --- PRI1 --- Asterisk --- PRI2 --- PBX I have done dozens and dozens of this type of implementation, sometimes you have to be very creative, but I have never failed. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Transfering Calls back on the same PRI
On Friday 17 October 2008 17:38, Steve Totaro wrote: I would engineer the system so that Asterisk is in the middle rather than the far end. Is there a reason why you don't want to or cannot do that? TELCO --- PRI1 --- Asterisk --- PRI2 --- PBX Steve, I have also done this same method in the past. In this case the number of PRI's entering the PBX far outweigh the number of PRI's in the Asterisk server, so it is not an option. I tried to simplify the example. Any other suggestions ? Ron -- Ron Joffe Siena Tech, Inc. 3319 Willow Glen Drive Oak Hill, VA 20171 (919) 928-0404 ___ -- 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
[asterisk-users] transfering calls
Hello folks! I've added a tT statement in my extensions (softphones and hardphones) so both the caller and calling parts can transfer calls (the strange thing is that some extensions end up only transfering the calls they originated, even if the config is the same of all other working extensions, which are able to transfer both placed and received calls). However, my trouble comes with the PSTN. I've added a t on the POTS- incoming s extension, so when it dial an extension, that extension can transfer the PSTN to someone else. Similarly, I've add a T to the outgoing calls Dial, so I can transfer a call to another extension after I place it. However, what happens is this: when I transfer a PSTN call to an extension, the PSTN gains the ability to transfer it! How I can prevent that to happen? Cheers, Francis ___ --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
[Asterisk-Users] Transfering calls. Dial plan
Hi there. We have lots of internal phones and we use the transfer option very often. But how do I set up the dial plan so that when a user transfers the call to someone else and that person is unavailable/busy (etc), the call returns to the user after a couple of seconds or so. I was thinking about using the CallerIdNum but the callerid is not always the number of the phone that transfers the phone. Sometimes its the other party of the converstation. This is basicly what i want to do External User calls in - One of us (person A) answers - Transfer call to correct person(B) - If person B unavailable, transfer back to person A. Thanks. ___ --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] transfering calls, no ringing sent to caller
I'm still trying to get this to work. I tried downgrading my Cisco phones to an earlier SIP image from cisco, thinking that might be the problem. Not it. Currently running SIP 7.5. I noticed when I use the transfer feature, a ZOMBIE appears on the channel, and the caller I'm transfering an extension or number to, doesn't hear any ringing. When using blind transfer, no zombie, and the ringing works! Anybody have any ideas how I can resolve this? Here's the console output. I was runninsg asterisk-1.2.0-beta1, but have upgraded to cvs head twice in hopes to get this problem resolved. Still no luck. -- Accepting call from '2532612594' to '2180650' on channel 0/1, span 1 -- Executing Dial(Zap/1-1, SIP/120|20|t) in new stack -- Called 120 -- SIP/120-ccab is ringing -- SIP/120-ccab answered Zap/1-1 -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Executing NoOp(SIP/120-841d, ) in new stack -- Executing Goto(SIP/120-841d, intern-post|500|1) in new stack -- Goto (intern-post,500,1) -- Executing Dial(SIP/120-841d, SIP/500|20|tr) in new stack -- Called 500 -- SIP/500-bdae is ringing -- Stopped music on hold on Zap/1-1 == Spawn extension (local, 2180650, 1) exited non-zero on 'SIP/120-841dZOMBIE' -- Channel 0/1, span 1 got hangup request == Spawn extension (intern-post, 500, 1) exited non-zero on 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (intern-post, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' On Wed, 28 Sep 2005, Jeremy Koski wrote: This worked for me in earlier versions, prior to 1.2.0-beta1. When I transfer an incoming caller to another number or extension, the caller does not hear any ringing. I have tried to generate the ringing using the r feature from the dial command with no luck. If I specify m instead of r, the caller hears the onhold music. I do hear the ringing on my end until I hit the transfer button. I've tried several combinations, but nothing seems to work. Any thoughts on how I can get this issue resolved? I'm using Cisco 7960 phones with asterisk-1.2.0-beta1. Thank you in advance! ___ --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
[Asterisk-Users] transfering calls, no ringing sent to caller
This worked for me in earlier versions, prior to 1.2.0-beta1. When I transfer an incoming caller to another number or extension, the caller does not hear any ringing. I have tried to generate the ringing using the r feature from the dial command with no luck. If I specify m instead of r, the caller hears the onhold music. I do hear the ringing on my end until I hit the transfer button. I've tried several combinations, but nothing seems to work. Any thoughts on how I can get this issue resolved? I'm using Cisco 7960 phones with asterisk-1.2.0-beta1. Thank you in advance! ___ --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
[Asterisk-Users] Transfering calls
Hello,
I'm new to asterisk and I'm really amaced of this software.
I've installed asterisk to take calls via SIP and Capi, all calls get
forwarded to a specific phone. Now I've to transfer some calls to other
phones. That workes already, but only this way, that I can transfer the
call without contacting the person on the new phone, for example to tell
him about the person that waits on the other line. So what I like to do
is the following:
1. A call comes in, either via SIP or Capi.
2. I take the call.
3. When I need to forward the call to an internal or an external number,
I'd like to park the call from outside and call the new phone. I'd
would be cool to play some music for the person who has to wait.
4. A person on the new phne hopefuly answers my call, I tell something
about the caller from outside, and after i hang up the waiting caller
from outside gets transfered automaticly to the new phone.
Is this possible with asterisk? If yes, how do I need to configure it
via the extensions.conf? Or is it not a problem of asterisk but a
problem of my IP phones (we use some Grandstreams and a Snom).
Here is my actual extensions.conf. Is this file OK in general or do I
have to change something (remember that I'm still a newbie :)). Can I
make some thing better? How can I setup the transfer stuff? Can anyone
give me an example or point me in the right direction please?
-
[general]
static=yes
writeprotect=no
;[globals]
;CONSOLE = Console/dsp
;IAXINFO = guest
;TRUNK = Zap/g2
;TRUNKMSD = 1
[local]
exten = 300,1,SetLanguage(de)
exten = 300,2,VoicemailMain()
exten = 300,3,Hangup()
exten = 301,1,Ringing()
exten = 301,2,Dial(SIP/301,30,t)
exten = 301,3,Congestion()
exten = 301,4,Busy()
exten = 301,5,Hangup()
exten = 302,1,Ringing()
exten = 302,2,Dial(SIP/302,30,t)
exten = 302,3,Congestion()
exten = 302,4,Busy()
exten = 302,5,Hangup()
exten = 303,1,Ringing()
exten = 303,2,Dial(SIP/303,30,t)
exten = 303,3,Congestion()
exten = 303,4,Busy()
exten = 303,5,Hangup()
[sip-in]
exten = anruf,1,Setlanguage(de)
exten = anruf,2,Ringing()
exten = anruf,3,Dial(SIP/302,30,t)
exten = anruf,4,Congestion()
;exten = anruf,3,Voicemail(302)
exten = anruf,5,Busy()
exten = anruf,6,Hangup()
exten = xyz,1,Goto(anruf,1)
[capi-in]
exten = isdn,1,Setlanguage(de)
exten = isdn,2,Ringing()
exten = isdn,3,Dial(SIP/302,30,t)
;exten = isdn,4,Voicemail(302)
exten = zyx,1,Goto(isdn,1)
[dialout]
include = local
exten = _0.,1,SetCallerID(zyx)
exten = _0.,2,Dial(CAPI/contr1/${EXTEN:1})
exten = _0.,3,Congestion()
exten = _0.,4,Busy()
exten = _0.,5,Hangup()
exten = _1.,1,SetCallerPres(some testing stuff)
exten = _1.,2,Dial(CAPI/contr1/${EXTEN:1})
exten = _1.,3,Congestion()
exten = _1.,4,Busy()
exten = _1.,5,Hangup()
exten = _2.,1,SetCallerID(zyx)
exten = _2.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tTr)
exten = _2.,3,Congestion()
exten = _2.,4,Busy()
exten = _2.,5,Hangup()
-
Best regards,
Schoeppi
--
Christian Schoepplein chris at schoeppi.net
Manage your communication: http://www.otrs.com
Linux for the blind: http://www.blinux.suse.de
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Re: [Asterisk-Users] Transfering calls or using any feature
Try taking out the capital T, having both causes problems in some configurations. Note that there are no defaults for features, you need to uncomment the entries in features.conf to activate the them. /M Anton Krall wrote: Guys, this is puzzling. Seems I cant use any of the feautes (call transfer, record call, etc) defined in features.conf when a call comes in thru zap and I answer it on hardphones... Although I CAN use them when Im the one that originates the call, when received I just cant. My dialplan includes wtWT on all Dial cmds just to be sure but it doesn't seem to be working. Any pointers? ___ 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
[Asterisk-Users] Transfering calls or using any feature
Guys, this is puzzling. Seems I cant use any of the feautes (call transfer, record call, etc) defined in features.conf when a call comes in thru zap and I answer it on hardphones... Although I CAN use them when Im the one that originates the call, when received I just cant. My dialplan includes wtWT on all Dial cmds just to be sure but it doesn't seem to be working. Any pointers? ___ 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] Transfering calls or using any feature
Guys, this is puzzling. Seems I cant use any of the feautes (call transfer, record call, etc) defined in features.conf when a call comes in thru zap and I answer it on hardphones... Although I CAN use them when Im the one that originates the call, when received I just cant. My dialplan includes wtWT on all Dial cmds just to be sure but it doesn't seem to be working. Any pointers? ___ 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] Transfering Calls
Try transfer and then send I believe. Thanks, Steve Totaro [EMAIL PROTECTED] www.totarotechnologies.com - Original Message - From: steve szmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 5:14 PM Subject: Re: [Asterisk-Users] Transfering Calls On Monday 25 October 2004 04:43 pm, [EMAIL PROTECTED] wrote: I have tried that on the GrandStream Budgetone phones and the transfer does not work on them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, October 25, 2004 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Transfering Calls Brian J. Rathman wrote: I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. The t and T options to Dial() provide # transfers. Use the transfer function of your SIP phone and don't use # transfers. I think the problem is the version of firmware. I used to be able to transfer fine but now I can't. On the other hand a lag I used to have is now gone. My V is 3.52. Using the hard Transfer button don't work, but using the softbutton does. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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
[Asterisk-Users] Transfering Calls
I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. Thanks, Brian ___ 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] Transfering Calls
Brian J. Rathman wrote: I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. The t and T options to Dial() provide # transfers. Use the transfer function of your SIP phone and don't use # transfers. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ 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] Transfering Calls
I have tried that on the GrandStream Budgetone phones and the transfer does not work on them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, October 25, 2004 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Transfering Calls Brian J. Rathman wrote: I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. The t and T options to Dial() provide # transfers. Use the transfer function of your SIP phone and don't use # transfers. ___ 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] Transfering Calls
On Monday 25 October 2004 04:43 pm, [EMAIL PROTECTED] wrote: I have tried that on the GrandStream Budgetone phones and the transfer does not work on them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, October 25, 2004 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Transfering Calls Brian J. Rathman wrote: I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. The t and T options to Dial() provide # transfers. Use the transfer function of your SIP phone and don't use # transfers. I think the problem is the version of firmware. I used to be able to transfer fine but now I can't. On the other hand a lag I used to have is now gone. My V is 3.52. Using the hard Transfer button don't work, but using the softbutton does. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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
