Re: [asterisk-users] Transfers from Queue Calls
A number of our clients has such issues. What we suggest for escalation is to do a blind transfer to a second-level queue, so that the logging is correct and even if second-line support cannot handle the call immediately, you get the functionality and the logging. Just my two euro cents, l. 2009/10/6 Darrin Henshaw darrin.aster...@gmail.com Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client wanted someone in particular, or maybe an escalation(we are a helpdesk). My problem is that the second part of the conversation after the transfer is not logged in the queue_log. Now this is by design from what I've found out, but we want the second part of the conversation to be recorded in the queue_log as well, for stats reporting for reviews of employee performance. Is anyone aware of a relatively easy way of implementing this? Whether it's by a patch or something else? Basically something similar to audiohook_inherit, which we use to allow mixmonitor to continue recording the call after it's been transferred. I've looked around, but haven't found anything. Thanks. Cheers, Darrin Henshaw -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Transfers from Queue Calls
Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client wanted someone in particular, or maybe an escalation(we are a helpdesk). My problem is that the second part of the conversation after the transfer is not logged in the queue_log. Now this is by design from what I've found out, but we want the second part of the conversation to be recorded in the queue_log as well, for stats reporting for reviews of employee performance. Is anyone aware of a relatively easy way of implementing this? Whether it's by a patch or something else? Basically something similar to audiohook_inherit, which we use to allow mixmonitor to continue recording the call after it's been transferred. I've looked around, but haven't found anything. Thanks. Cheers, Darrin Henshaw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Transfers from Queue Calls
On 091006 1249, Darrin Henshaw wrote: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client wanted someone in particular, or maybe an escalation(we are a helpdesk). My problem is that the second part of the conversation after the transfer is not logged in the queue_log. Have you considered using CDR instead? Generally, you get 2 CDR records per call, one before the transfer and the second after. YMMV depending on transfer method and agent phone technology. -kkm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Transfers on an inter-PBX PRI link
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension
103
Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension
104
Step3 fails and extension 103 is reconnected to 101.
Why is Step3 failing and how could I change my setup so the transfer succeeds?
As a side question, I'd like to know if I could free the unnecessary zap
channels created in Steps 1 and 2.
On analog channels I could SendDTMF(${EXTEN}). I don't know how to do that on a
digital pri line and if it requires that the legacy PBX be compatible.
Anyway, I'm not too worried about freeing the PRI channels. I just want Step3
to work.
Is it possible, somehow?
Thanks in advance,
Vieri
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Re: [asterisk-users] Transfers on an inter-PBX PRI link
On Mon, Mar 16, 2009 at 8:49 AM, Vieri rentor...@yahoo.com wrote:
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension
103
Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension
104
Step3 fails and extension 103 is reconnected to 101.
Why is Step3 failing and how could I change my setup so the transfer succeeds?
As a side question, I'd like to know if I could free the unnecessary zap
channels created in Steps 1 and 2.
On analog channels I could SendDTMF(${EXTEN}). I don't know how to do that on
a digital pri line and if it requires that the legacy PBX be compatible.
Anyway, I'm not too worried about freeing the PRI channels. I just want
Step3 to work.
Is it possible, somehow?
Thanks in advance,
Vieri
Relevant parts of your dialplan, tech.conf, and debug info is probably
the only way to really help you besides making wild guesses.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Transfers on AgentLogin()
Mark Hamilton wrote: a) How can I make it so #2 doesn't have to be exceptionally fast, and maybe get a second of delay in there permitted? ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation (default is 500 ms) -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Transfers on AgentLogin()
Hi James,Thank you very much for a detailed reply. (Matt, sorry about earlier, I totally missed the part you said about the t option)To answer, yes the Queue command does have t and T passed to it. This is how I tested it. Agent1 is on eyeBeam and he's the one who will need to do an attended transfer to a queue. So, let's say the shortcode to the queue is 3. Agent1 gets a call, presses the # (even though the transfer sequence is set to #2.. immediately, Agent1 heard "Transfer", which means just the # was enough to put it in the transfer mode) and the minute Agent1 presses 3, it's a blind transfer. canreinvite=no and so dtmf=auto. It doesn't seem to be picking up the feature codes set in features.conf for some reason. So # is doing the transfer, even though the only thing uncommented in features.conf was atndxfer, which was set to *2 and then to #2 since *2 was doing a hangup (the hangup sequence for agentlogin). dtmfmode couldn't be set to info because eyeBeam is used by Agent1 and DTMF wasn't being recognized when the agent was trying to login to the queue.[1013]type=friendqualify=yesnat=yeshost=dynamicdtmfmode=autocontext=manilacanreinvite=nocallerid=Agent 1013call-limit=10Please helpThanks! Original Message Subject: Re: [asterisk-users] Transfers on AgentLogin() From: "James Sneeringer" [EMAIL PROTECTED] Date: Fri, September 05, 2008 10:57 pm To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Since AgentLogin() essentially keeps a channel to the agent open all the time, a normal SIP transfer will do exactly as you say. That is, it will try to send the agent's login session into queue, which isn't what you want. As Matt suggested, you need to pass the "t" option to the Queue() application. This will let your agents perform a DTMF transfer using the codes defined in features.conf. The agent basically dials a short code while talking to the caller. Asterisk intercepts it, and then prompts the agent for the extension to transfer the call to. Look in features.conf for more information. Fair warning, I have never needed to use this feature, so I can't attest to exactly how it behaves. We use dynamic agent logins, so we've never had to deal with AgentLogin(). This allows us to do normal SIP transfers. Also, you will probably have to do one of two things in your sip.conf. One, set "canreinvite" to "no" to keep Asterisk in the call path, that way it can intercept the DTMF tones. Or, two, set "dtmfmode" to "info", so that DTMF tones are converted to SIP INFO messages, which Asterisk will see. At least, that's how I think it works. :) -James On Sun, Aug 31, 2008 at 3:15 PM, Mark Hamilton [EMAIL PROTECTED] wrote: I've tried the regular, xfer button on xlite, dial 100 (to transfer to the queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(), so it transfers the full persistent connection to the queue instead of the call itself and this causes the transferring agent to logout. Either that, or I'm doing something wrong. There is no documentation out there so I don't know how it would work for AgentLogin(). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Matt Riddell Sent: August 30, 2008 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers on AgentLogin() What did you try and how did it fail? Are you using the t option in queue? ___ -- 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 ___ -- 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] Transfers on AgentLogin()
What do you get when you type show features? On 9/6/08, Mark Hamilton [EMAIL PROTECTED] wrote: Hi James, Thank you very much for a detailed reply. (Matt, sorry about earlier, I totally missed the part you said about the t option) To answer, yes the Queue command does have t and T passed to it. This is how I tested it. Agent1 is on eyeBeam and he's the one who will need to do an attended transfer to a queue. So, let's say the shortcode to the queue is 3. Agent1 gets a call, presses the # (even though the transfer sequence is set to #2.. immediately, Agent1 heard Transfer, which means just the # was enough to put it in the transfer mode) and the minute Agent1 presses 3, it's a blind transfer. canreinvite=no and so dtmf=auto. It doesn't seem to be picking up the feature codes set in features.conf for some reason. So # is doing the transfer, even though the only thing uncommented in features.conf was atndxfer, which was set to *2 and then to #2 since *2 was doing a hangup (the hangup sequence for agentlogin). dtmfmode couldn't be set to info because eyeBeam is used by Agent1 and DTMF wasn't being recognized when the agent was trying to login to the queue. [1013] type=friend qualify=yes nat=yes host=dynamic dtmfmode=auto context=manila canreinvite=no callerid=Agent 1013 call-limit=10 Please help Thanks! Original Message Subject: Re: [asterisk-users] Transfers on AgentLogin() From: James Sneeringer [EMAIL PROTECTED] Date: Fri, September 05, 2008 10:57 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Since AgentLogin() essentially keeps a channel to the agent open all the time, a normal SIP transfer will do exactly as you say. That is, it will try to send the agent's login session into queue, which isn't what you want. As Matt suggested, you need to pass the t option to the Queue() application. This will let your agents perform a DTMF transfer using the codes defined in features.conf. The agent basically dials a short code while talking to the caller. Asterisk intercepts it, and then prompts the agent for the extension to transfer the call to. Look in features.conf for more information. Fair warning, I have never needed to use this feature, so I can't attest to exactly how it behaves. We use dynamic agent logins, so we've never had to deal with AgentLogin(). This allows us to do normal SIP transfers. Also, you will probably have to do one of two things in your sip.conf. One, set canreinvite to no to keep Asterisk in the call path, that way it can intercept the DTMF tones. Or, two, set dtmfmode to info, so that DTMF tones are converted to SIP INFO messages, which Asterisk will see. At least, that's how I think it works. :) -James On Sun, Aug 31, 2008 at 3:15 PM, Mark Hamilton [EMAIL PROTECTED] wrote: I've tried the regular, xfer button on xlite, dial 100 (to transfer to the queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(), so it transfers the full persistent connection to the queue instead of the call itself and this causes the transferring agent to logout. Either that, or I'm doing something wrong. There is no documentation out there so I don't know how it would work for AgentLogin(). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: August 30, 2008 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers on AgentLogin() What did you try and how did it fail? Are you using the t option in queue? ___ -- 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 -- Sent from Gmail for mobile | mobile.google.com Matt Riddell Director VentureVoIP ___ -- 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] Transfers on AgentLogin()
Hi Matt, I guess I needed to dial the #2 REAL FAST to get the transfer sequence and show features showed nothing because I was reloading, not restarting. After I figured out the restart: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ##9 Attended Transfer #2 One Touch Monitor Disconnect Call * * Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 700 Parking context : default Parked call extensions: 701-720 Now, on the eyebeam when I'm on the persistent agentlogin() and the call comes in, I have to do #2 REAL FAST, and it says Transfer, I type 2 and it transfers to an external call and I can talk to the agent on 2, while the caller hears the hold music. So, I guess the only two things I need to figure out now are: a) How can I make it so #2 doesn't have to be exceptionally fast, and maybe get a second of delay in there permitted? b) After I start the transfer and talk to the other agent about the caller I'm about to transfer, how do I 1) patch the caller into the call with me and the other agent, and then when they start getting friendly and I want to leave, how do I leave that call? Thanks a lot guys! PS: Totally unrelated, but if this agent's internet goes down, somehow the queue still keeps him logged in for atleast a few minutes. When agent gets internet back and tries to log back in, it says already logged in unless it automatically falls off or someone force logs them out. How can I solve? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: September 6, 2008 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers on AgentLogin() What do you get when you type show features? On 9/6/08, Mark Hamilton [EMAIL PROTECTED] wrote: Hi James, Thank you very much for a detailed reply. (Matt, sorry about earlier, I totally missed the part you said about the t option) To answer, yes the Queue command does have t and T passed to it. This is how I tested it. Agent1 is on eyeBeam and he's the one who will need to do an attended transfer to a queue. So, let's say the shortcode to the queue is 3. Agent1 gets a call, presses the # (even though the transfer sequence is set to #2.. immediately, Agent1 heard Transfer, which means just the # was enough to put it in the transfer mode) and the minute Agent1 presses 3, it's a blind transfer. canreinvite=no and so dtmf=auto. It doesn't seem to be picking up the feature codes set in features.conf for some reason. So # is doing the transfer, even though the only thing uncommented in features.conf was atndxfer, which was set to *2 and then to #2 since *2 was doing a hangup (the hangup sequence for agentlogin). dtmfmode couldn't be set to info because eyeBeam is used by Agent1 and DTMF wasn't being recognized when the agent was trying to login to the queue. [1013] type=friend qualify=yes nat=yes host=dynamic dtmfmode=auto context=manila canreinvite=no callerid=Agent 1013 call-limit=10 Please help Thanks! Original Message Subject: Re: [asterisk-users] Transfers on AgentLogin() From: James Sneeringer [EMAIL PROTECTED] Date: Fri, September 05, 2008 10:57 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Since AgentLogin() essentially keeps a channel to the agent open all the time, a normal SIP transfer will do exactly as you say. That is, it will try to send the agent's login session into queue, which isn't what you want. As Matt suggested, you need to pass the t option to the Queue() application. This will let your agents perform a DTMF transfer using the codes defined in features.conf. The agent basically dials a short code while talking to the caller. Asterisk intercepts it, and then prompts the agent for the extension to transfer the call to. Look in features.conf for more information. Fair warning, I have never needed to use this feature, so I can't attest to exactly how it behaves. We use dynamic agent logins, so we've never had to deal with AgentLogin(). This allows us to do normal SIP transfers. Also, you will probably have to do one of two things in your sip.conf. One, set canreinvite to no to keep Asterisk in the call path, that way it can intercept the DTMF tones. Or, two, set dtmfmode to info, so that DTMF tones are converted to SIP INFO messages, which Asterisk will see. At least, that's how I think it works. :) -James On Sun, Aug 31, 2008 at 3:15 PM, Mark Hamilton [EMAIL PROTECTED] wrote: I've tried the regular, xfer button on xlite, dial 100 (to transfer to the queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(), so it transfers the full
Re: [asterisk-users] Transfers on AgentLogin()
So, nobody? How is Asterisk vying to become a bigtime key player in PBX systems when some things are not documented, and one cannot get help on a mailing list or irc (maybe because people don't know themselves)? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 31, 2008 4:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Transfers on AgentLogin() I've tried the regular, xfer button on xlite, dial 100 (to transfer to the queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(), so it transfers the full persistent connection to the queue instead of the call itself and this causes the transferring agent to logout. Either that, or I'm doing something wrong. There is no documentation out there so I don't know how it would work for AgentLogin(). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: August 30, 2008 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers on AgentLogin() What did you try and how did it fail? Are you using the t option in queue? On 8/30/08, Mark Hamilton [EMAIL PROTECTED] wrote: So, no answers or is this thread going to remain unanswered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Transfers on AgentLogin() Oh, by the way, the agent who will be doing the assisted transfer will be using eyebeam. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 5:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transfers on AgentLogin() Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. -- Sent from Gmail for mobile | mobile.google.com Matt Riddell Director VentureVoIP ___ -- 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 ___ -- 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 ___ -- 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] Transfers on AgentLogin()
Since AgentLogin() essentially keeps a channel to the agent open all the time, a normal SIP transfer will do exactly as you say. That is, it will try to send the agent's login session into queue, which isn't what you want. As Matt suggested, you need to pass the t option to the Queue() application. This will let your agents perform a DTMF transfer using the codes defined in features.conf. The agent basically dials a short code while talking to the caller. Asterisk intercepts it, and then prompts the agent for the extension to transfer the call to. Look in features.conf for more information. Fair warning, I have never needed to use this feature, so I can't attest to exactly how it behaves. We use dynamic agent logins, so we've never had to deal with AgentLogin(). This allows us to do normal SIP transfers. Also, you will probably have to do one of two things in your sip.conf. One, set canreinvite to no to keep Asterisk in the call path, that way it can intercept the DTMF tones. Or, two, set dtmfmode to info, so that DTMF tones are converted to SIP INFO messages, which Asterisk will see. At least, that's how I think it works. :) -James On Sun, Aug 31, 2008 at 3:15 PM, Mark Hamilton [EMAIL PROTECTED] wrote: I've tried the regular, xfer button on xlite, dial 100 (to transfer to the queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(), so it transfers the full persistent connection to the queue instead of the call itself and this causes the transferring agent to logout. Either that, or I'm doing something wrong. There is no documentation out there so I don't know how it would work for AgentLogin(). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: August 30, 2008 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers on AgentLogin() What did you try and how did it fail? Are you using the t option in queue? ___ -- 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] Transfers on AgentLogin()
I've tried the regular, xfer button on xlite, dial 100 (to transfer to the queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(), so it transfers the full persistent connection to the queue instead of the call itself and this causes the transferring agent to logout. Either that, or I'm doing something wrong. There is no documentation out there so I don't know how it would work for AgentLogin(). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: August 30, 2008 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers on AgentLogin() What did you try and how did it fail? Are you using the t option in queue? On 8/30/08, Mark Hamilton [EMAIL PROTECTED] wrote: So, no answers or is this thread going to remain unanswered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Transfers on AgentLogin() Oh, by the way, the agent who will be doing the assisted transfer will be using eyebeam. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 5:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transfers on AgentLogin() Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. -- Sent from Gmail for mobile | mobile.google.com Matt Riddell Director VentureVoIP ___ -- 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 ___ -- 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] Transfers on AgentLogin()
What did you try and how did it fail? Are you using the t option in queue? On 8/30/08, Mark Hamilton [EMAIL PROTECTED] wrote: So, no answers or is this thread going to remain unanswered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Transfers on AgentLogin() Oh, by the way, the agent who will be doing the assisted transfer will be using eyebeam. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 5:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transfers on AgentLogin() Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. -- Sent from Gmail for mobile | mobile.google.com Matt Riddell Director VentureVoIP ___ -- 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] Transfers on AgentLogin()
So, no answers or is this thread going to remain unanswered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Transfers on AgentLogin() Oh, by the way, the agent who will be doing the assisted transfer will be using eyebeam. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 5:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transfers on AgentLogin() Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. ___ -- 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
[asterisk-users] Transfers on AgentLogin()
Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. ___ -- 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] Transfers on AgentLogin()
Oh, by the way, the agent who will be doing the assisted transfer will be using eyebeam. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 5:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transfers on AgentLogin() Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. ___ -- 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
[asterisk-users] transfers only work when voicemail enabled
Hi all, when enabling blind and attended transfers in features.conf, these only seem to work when I enable voicemail for a particular user. How can this be? Can I have transferrring without voicemail? Using Asterisk 1.4 by the way. Thank you! Bart ___ -- 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] transfers only work when voicemail enabled
Bart Coninckx wrote: Hi all, when enabling blind and attended transfers in features.conf, these only seem to work when I enable voicemail for a particular user. How can this be? Can I have transferrring without voicemail? Using Asterisk 1.4 by the way. Thank you! Bart I think some clarification is necessary here. What do you mean by enable voicemail? Do you mean that you add a Voicemail() application call to the Dialplan? I don't see how that could make a difference regarding whether transfers are allowed. Transferring should be allowable just by adding either the 't' or 'T' flags to the options for Dial(). Mark Michelson ___ -- 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] transfers only work when voicemail enabled
Hi all, when enabling blind and attended transfers in features.conf, these only seem to work when I enable voicemail for a particular user. How can this be? Can I have transferrring without voicemail? Using Asterisk 1.4 by the way. Thank you! Bart I think some clarification is necessary here. What do you mean by enable voicemail? Do you mean that you add a Voicemail() application call to the Dialplan? I don't see how that could make a difference regarding whether transfers are allowed. Transferring should be allowable just by adding either the 't' or 'T' flags to the options for Dial(). Mark Michelson Hi Mark, yes, I'm sorry, I should have been more clear about this: I'm referring to the hasvoicemail setting in the users.conf file. When this is set to no, transferring does not work. When set to yes, it does. Both t and T are added to my Dial commands, thank you, Bart ___ -- 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
[asterisk-users] Transfers with TE12xp
Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 - ___ -- 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] Transfers with TE12xp
More info about the problem. This occurs, when I try to transfer using the *2 funcionality into aterisk Thanks 2008/6/16 voip crazy [EMAIL PROTECTED]: Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 - ___ -- 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] Transfers with TE12xp
On Mon, Jun 16, 2008 at 6:39 AM, voip crazy [EMAIL PROTECTED] wrote: Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 I don't see anything obviously wrong with the above. How about some verbose output from the Asterisk CLI? If that doesn't shed some light on it, how about pri debug span 1 output? Thanks, Steve ___ -- 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] transfers and CDR
Hi everybody, A question, how do I follow a call that is transferred? is the any event or something in the CDR that would let me find all the call sequence? Thanks Rodrigo ___ --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] Transfers - No ringback or moh
Hi Mike, Hi all, really works. ;-) But that can not be the solution for the future? :-) Can it? I think there should be an ANSWER() implimented in the Transfer function to prevent this problem ... Or does anybody have other ideas? greetings, Martin - Original Message - From: Mike Dawson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 26, 2006 4:32 PM Subject: Re: [asterisk-users] Transfers - No ringback or moh I get round this bug by replacing: exten = X,1,Dial(sip/blah) with: exten = X,1,Answer exten = X,n,Dial(sip/blah) It means the call is in an answered state before it starts ringing but it doesn't seem to cause any major problems. Mike Martin Schrott - Thinking-Systems wrote: Hi all, I cannot exactly reproduce your problems, but I can tell you, what problem we have on this topic: a calles b. b takes the call and can speak to a. b sets up a attendend transfer (via the softkey configured in asterisk) to c and hears ringing. a hears music on hold. b hears ringing if c answeres and b hanges up, everything is fine. now the problem: if b hangs up, before c has answered (during ringing) a will loose the connection and also be hanged up. I think this should not happen! The transfer should automatically be changed to blind and a should get the ringing played back instead of b. Hope, you can understand my problem and may have any ideas or thoughts. Greetings and Thanks, Martin ___ --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] Transfers - No ringback or moh
I get round this bug by replacing: exten = X,1,Dial(sip/blah) with: exten = X,1,Answer exten = X,n,Dial(sip/blah) It means the call is in an answered state before it starts ringing but it doesn't seem to cause any major problems. Mike Martin Schrott - Thinking-Systems wrote: Hi all, I cannot exactly reproduce your problems, but I can tell you, what problem we have on this topic: a calles b. b takes the call and can speak to a. b sets up a attendend transfer (via the softkey configured in asterisk) to c and hears ringing. a hears music on hold. b hears ringing if c answeres and b hanges up, everything is fine. now the problem: if b hangs up, before c has answered (during ringing) a will loose the connection and also be hanged up. I think this should not happen! The transfer should automatically be changed to blind and a should get the ringing played back instead of b. Hope, you can understand my problem and may have any ideas or thoughts. Greetings and Thanks, Martin ___ --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] Transfers - No ringback or moh
Hi all, I cannot exactly reproduce your problems, but I can tell you, what problem we have on this topic: a calles b. b takes the call and can speak to a. b sets up a attendend transfer (via the softkey configured in asterisk) to c and hears ringing. a hears music on hold. b hears ringing if c answeres and b hanges up, everything is fine. now the problem: if b hangs up, before c has answered (during ringing) a will loose the connection and also be hanged up. I think this should not happen! The transfer should automatically be changed to blind and a should get the ringing played back instead of b. Hope, you can understand my problem and may have any ideas or thoughts. Greetings and Thanks, Martin ___ --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] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)
Hi all, Here is the situation: A call comes in to an Alcatel PBX and it sends it to an E1 on * , this * either sends the call to a VoIP extension or needs to forward it to an extension back on the Alcatel, but WITHOUT using another slot of the E1 (no tromboning or hairpinning). I've read you can do this with 2b channel transfers implemented on 5ESS, and also on QSIG. I know Matthew Fredrickson did it on * (I think he programmed it for *) I also know there is quite a bit of people pursuing this same goal, which is way important to lower the income barriers for * to enter the legacy world. Has anyone actually done it? I appreciate any input whatsoever, and if possible a sample of how to manage it on *.What to put on the extensions.conf to perform the transfer and any other files needed, thanks ___ --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] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)
On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote: Hi all, Here is the situation: A call comes in to an Alcatel PBX and it sends it to an E1 on * , this * either sends the call to a VoIP extension or needs to forward it to an extension back on the Alcatel, but WITHOUT using another slot of the E1 (no tromboning or hairpinning). I've read you can do this with 2b channel transfers implemented on 5ESS, and also on QSIG. I know Matthew Fredrickson did it on * (I think he programmed it for *) I also know there is quite a bit of people pursuing this same goal, which is way important to lower the income barriers for * to enter the legacy world. Has anyone actually done it? I appreciate any input whatsoever, and if possible a sample of how to manage it on *.What to put on the extensions.conf to perform the transfer and any other files needed, Unfortunately, I have not implemented the Q.SIG version of 2b channel transfer, so for the time being you'll have to stick to hairpinning the legs of the call. The Q.SIG version is a little bit more complicated than some of the other versions. Matthew Fredrickson ___ --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] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)
Hi Matt, thanks for your answer, I guess it is still as you said a while back that you did it using 5ESS Can you share how you did in 5ESS? (a sample of the extensions.conf ) and what kind of switch you were connected to? I'm not sure if the Alcatel 4400 and the Nortel Meridian 11 supports 5ESS, but are willing to find out. thanks Manrique Matthew Fredrickson escribió: On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote: Hi all, Here is the situation: A call comes in to an Alcatel PBX and it sends it to an E1 on * , this * either sends the call to a VoIP extension or needs to forward it to an extension back on the Alcatel, but WITHOUT using another slot of the E1 (no tromboning or hairpinning). I've read you can do this with 2b channel transfers implemented on 5ESS, and also on QSIG. I know Matthew Fredrickson did it on * (I think he programmed it for *) I also know there is quite a bit of people pursuing this same goal, which is way important to lower the income barriers for * to enter the legacy world. Has anyone actually done it? I appreciate any input whatsoever, and if possible a sample of how to manage it on *.What to put on the extensions.conf to perform the transfer and any other files needed, Unfortunately, I have not implemented the Q.SIG version of 2b channel transfer, so for the time being you'll have to stick to hairpinning the legs of the call. The Q.SIG version is a little bit more complicated than some of the other versions. Matthew Fredrickson ___ --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] Transfers - No ringback or moh
I don't know why, but when doing transfers between Polycom phones, once the transferring party hits transfer a second time, to be removed from the call, User A no longer hears music on hold, or a ring back. Scenario. 1. User A dials User B. 2. User A and User B are connected. 3. User B hits the transfer soft key. User A gets music on hold. 4. User B dials user C. User C's phone rings, and user A continues to hear music on hold. 5. When User B presses the transfer soft key again to complete the transfer, the music on hold for User A stops. Question is, why? Doug a ___ --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] Transfers - No ringback or moh
Douglas Garstang wrote: 1. User A dials User B. 2. User A and User B are connected. 3. User B hits the transfer soft key. User A gets music on hold. 4. User B dials user C. User C's phone rings, and user A continues to hear music on hold. 5. When User B presses the transfer soft key again to complete the transfer, the music on hold for User A stops. Because user B just did an attended transfer. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Transfers - No ringback or moh
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: 1. User A dials User B. 2. User A and User B are connected. 3. User B hits the transfer soft key. User A gets music on hold. 4. User B dials user C. User C's phone rings, and user A continues to hear music on hold. 5. When User B presses the transfer soft key again to complete the transfer, the music on hold for User A stops. Because user B just did an attended transfer. And that's normal, for user A to just hear dead air? Doug. ___ --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] Transfers - No ringback or moh
Douglas Garstang wrote: -Original Message- And that's normal, for user A to just hear dead air? I have a Polycom IP501 sitting on my desk (Test phone): I call it with my Avaya phone pick up the ringing extension press transfer button (I hear hold music on the Avaya) I dial the voice mail extension on the Asterisk I press the transfer button again. Hold music stops and I hear Comedian Mail. So, is the dead air that you hear, the silence that you would get when the hold music stops and both parties have been bridged, or no audio is passes at all? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Transfers - No ringback or moh
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: -Original Message- And that's normal, for user A to just hear dead air? I have a Polycom IP501 sitting on my desk (Test phone): I call it with my Avaya phone pick up the ringing extension press transfer button (I hear hold music on the Avaya) I dial the voice mail extension on the Asterisk I press the transfer button again. Hold music stops and I hear Comedian Mail. So, is the dead air that you hear, the silence that you would get when the hold music stops and both parties have been bridged, or no audio is passes at all? I don't know... all I know is that when user C starts to ring, and user B has dropped from the call, the music on hold stops for user A, until user C answers. I would have expected User A to hear ringing at this point. Doug. ___ --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] Transfers - No ringback or moh
Douglas Garstang wrote: I don't know... all I know is that when user C starts to ring, and user B has dropped from the call, the music on hold stops for user A, until user C answers. I would have expected User A to hear ringing at this point. Then they need to do a blind transfer. Transfer button, Blind (Soft button) Extensions or phone number to transfer too Depending on your digit map for the Polycom, you may have to press something after that. My digit map matches against 4 digit extensions that we use internally, 7 and 10 digit number starting with a 9 are also matched. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Transfers - No ringback or moh
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: I don't know... all I know is that when user C starts to ring, and user B has dropped from the call, the music on hold stops for user A, until user C answers. I would have expected User A to hear ringing at this point. Then they need to do a blind transfer. Transfer button, Blind (Soft button) Extensions or phone number to transfer too Depending on your digit map for the Polycom, you may have to press something after that. My digit map matches against 4 digit extensions that we use internally, 7 and 10 digit number starting with a 9 are also matched. Doug, The transfer soft button can do both attended and non attended transfers. If user B presses the transfer soft button before user C picks up, it's an unattended transfer. If user B presses the transfer soft button after user C has answered, then it's an attended transfer. Doing a blind/unattended transfer isn't going to make any difference, as the ringback (or lack of it) to user A, occurs as soon as user B presses the transfer soft button a second time. Don't see what the Polycom digit map has to do with it. Doug. ___ --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] Transfers - No ringback or moh
Douglas Garstang wrote: -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Doug, The transfer soft button can do both attended and non attended transfers. If user B presses the transfer soft button before user C picks up, it's an unattended transfer. If user B presses the transfer soft button after user C has answered, then it's an attended transfer. I'm not able reproduct this. If I don't select Blind before entering the extension or number when transferring to user C, it only offers a cancel or split. Once user C answers, then I'm offered the Transfer soft button. Don't see what the Polycom digit map has to do with it. Not wanting to press yet another button. People (Users that is) seem to think, pressing less is better. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Transfers - No ringback or moh
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: -Original Message- And that's normal, for user A to just hear dead air? I have a Polycom IP501 sitting on my desk (Test phone): I call it with my Avaya phone pick up the ringing extension press transfer button (I hear hold music on the Avaya) I dial the voice mail extension on the Asterisk I press the transfer button again. Hold music stops and I hear Comedian Mail. So, is the dead air that you hear, the silence that you would get when the hold music stops and both parties have been bridged, or no audio is passes at all? I don't think this is the same scenario. When you transfer to an Asterisk extension, ie voicemail, your not going to get a period of ring back as Asterisk will answer the call immediately. Douglas. ___ --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] Transfers - No ringback or moh
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Doug, The transfer soft button can do both attended and non attended transfers. If user B presses the transfer soft button before user C picks up, it's an unattended transfer. If user B presses the transfer soft button after user C has answered, then it's an attended transfer. I'm not able reproduct this. If I don't select Blind before entering the extension or number when transferring to user C, it only offers a cancel or split. Once user C answers, then I'm offered the Transfer soft button. Don't see what the Polycom digit map has to do with it. Not wanting to press yet another button. People (Users that is) seem to think, pressing less is better. Doug, as it turns out, the transfer button on the polycom and the transfer soft button, both behave in exactly the same way. If user B wants to transfer user A to user C, ATTENDED, user B simply presses the transfer button and waits for user C to pick up before pressing the transfer button again. If user B wants to transfer user A to user C, UNATTENDED, user B simply presses the transfer button and presses it again before user C picks up. In any case, when the phone of user C is ringing, user A does not hear a ring. Doug. ___ --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] Transfers - No ringback or moh
Douglas Garstang wrote: passes at all? I don't think this is the same scenario. When you transfer to an Asterisk extension, ie voicemail, your not going to get a period of ring back as Asterisk will answer the call immediately. In this example, I'm dialing to another extension on my desk. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Transfers - No ringback or moh
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: passes at all? I don't think this is the same scenario. When you transfer to an Asterisk extension, ie voicemail, your not going to get a period of ring back as Asterisk will answer the call immediately. In this example, I'm dialing to another extension on my desk. When the extension on your desk is ringing, after you have pressed transfer key a second time(soft or hard key), does the original caller still hear music on hold, or ringback or nothing? Douglas. ___ --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] Transfers - No ringback or moh
Douglas Garstang wrote: People (Users that is) seem to think, pressing less is better. Doug, as it turns out, the transfer button on the polycom and the transfer soft button, both behave in exactly the same way. What firmware? I'm running Bootrom 3.1.3, sip.ld 1.5.2 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Transfers - No ringback or moh
Douglas Garstang wrote: -Original Message- When the extension on your desk is ringing, after you have pressed transfer key a second time(soft or hard key), does the original caller still hear music on hold, or ringback or nothing? Following your example, pressing transfer once, entering the extension (Caller C's) does not yield a second transfer option until C answers. Pressing the button anyway, does not get a response from the phone. When (B) selects the initial transfer, I have Cancel, Name, Blind. (A) hears hold music. During the transfer and Before (C) answers, the phone options are Cancel, Split. Once C answers, I have, Hold, Cancel, Transfer, More Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Transfers - No ringback or moh
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: -Original Message- When the extension on your desk is ringing, after you have pressed transfer key a second time(soft or hard key), does the original caller still hear music on hold, or ringback or nothing? Following your example, pressing transfer once, entering the extension (Caller C's) does not yield a second transfer option until C answers. Pressing the button anyway, does not get a response from the phone. When (B) selects the initial transfer, I have Cancel, Name, Blind. (A) hears hold music. During the transfer and Before (C) answers, the phone options are Cancel, Split. Once C answers, I have, Hold, Cancel, Transfer, More We have SIP version 1.6.3. Polycom must have changed something... Doug. ___ --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] Transfers - No ringback or moh
On 7/24/06, Douglas Garstang [EMAIL PROTECTED] wrote: Douglas Garstang wrote: -Original Message- When the extension on your desk is ringing, after you have pressed transfer key a second time(soft or hard key), does the original caller still hear music on hold, or ringback or nothing? Following your example, pressing transfer once, entering the extension (Caller C's) does not yield a second transfer option until C answers. Pressing the button anyway, does not get a response from the phone. When (B) selects the initial transfer, I have Cancel, Name, Blind. (A) hears hold music. During the transfer and Before (C) answers, the phone options are Cancel, Split. Once C answers, I have, Hold, Cancel, Transfer, More We have SIP version 1.6.3. Polycom must have changed something... Doug. Not exactly. There's an option in the Polycom config to disallow unattended attended transfers. In other words, you do not have the option to press Transfer while you are getting in a RINGING progress state. Sounds Like one of you has that option enabled, the other has it disabled. I don't know exactly what the option is, but I've seen it before. Nathan ___ --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] transfers/parked calls + polycom 501
I am trying to get parked calls/transfers working on our polycom 501s + asterisk.The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call.Furthermore the # - 700 only works on incomming calls. If I dial out then try to transfer, the # - 700 doesn't do anything. Thanks[meetme-ext] exten = 600,1,MeetMe(1234|Mp|98765)[extentions] include = parkedcalls include = meetme-ext exten = _10XX,1,Dial(SIP/${EXTEN},20,tT) exten = _10XX,n,Answer exten = _10XX,n,VoiceMail([EMAIL PROTECTED]) exten = _10XX,n,Hangup()[voicemail] exten = _910XX,1,Wait(1) exten = _910XX,n,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])[local] include = extentions include = voicemail[incoming] ;exten =
s,1,Zapateller(nocallerid) exten = s,1,Answer exten = s,n,Wait(2) exten = s,n,Set(TIMEOUT(response)=15) exten = s,n,Background(intro) exten = s,n,WaitExten() exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup() exten = 0,1,Dial(${ATTENDANT},20) exten = 0,n,Playback(vm-nobodyavail) exten = 0,n,Hangup() exten = 1,1,Directory(voicemail,extentions,f) exten = 2,1,Directory(voicemail,extentions) include = meetme-ext include = extentions exten = i,1,Playback(vm-goodbye) exten = i,2,Hangup() exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup()[outbound] ignorepat = 9 include = parkedcalls exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1}) exten = _9XX,2,Congestion() exten = _9XX,102,Congestion() exten = _91900NXX,1,Congestion()
exten = _91976NXX,1,Congestion() exten = _91[123456789]XXNXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1}) exten = _91[123456789]XXNXX,2,Congestion() exten = _91[123456789]XXNXX,102,Congestion() exten = 9911,1,Dial(${OUTBOUNDTRUNK}/ww911) exten = 9411,1,Dial(${OUTBOUNDTRUNK}/ww411) exten = 0,1,Dial(${OUTBOUNDTRUNK}/ww0)[local-access] include = local include = outbound
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[Asterisk-Users] transfers using # in asterisk
Greetings fellow list members, I am using ABE and I am attempting to impliment transfers using "#". I am using both "T" and "t" as options in my Dial() command. I am attempting to hit "#" then enter another extension from my dialplan. I have tried this on both ends of the conversation and also tried hitting "#" again after entering the extension and still no luck. One end of the conversation is a SNOM 320, the other is an outside line. The transfer does not happen, I was wondering if anyone had any suggestions for me, perhaps something easily missed. I've looked at the wiki and I do have canreinvite set to no. Any help or ideas are much appreciated. Thank you, Frank Webb Inter Media Marketing Solutions ___ --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] transfers using # in asterisk
Any idea what version of Asterisk ABE is based on? PaulH - Original Message - From: Franklin Webb To: asterisk-users@lists.digium.com Sent: Friday, December 30, 2005 8:43 AM Subject: [Asterisk-Users] transfers using # in asterisk Greetings fellow list members, I am using ABE and I am attempting to impliment transfers using "#". I am using both "T" and "t" as options in my Dial() command. I am attempting to hit "#" then enter another extension from my dialplan. I have tried this on both ends of the conversation and also tried hitting "#" again after entering the extension and still no luck. One end of the conversation is a SNOM 320, the other is an outside line. The transfer does not happen, I was wondering if anyone had any suggestions for me, perhaps something easily missed. I've looked at the wiki and I do have canreinvite set to no. Any help or ideas are much appreciated. Thank you, Frank Webb Inter Media Marketing Solutions ___--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] transfers using # in asterisk
On Thu, 2005-12-29 at 16:43 -0500, Franklin Webb wrote: Greetings fellow list members, I am using ABE and I am attempting to impliment transfers using #. I am using both T and t as options in my Dial() command. I am attempting to hit # then enter another extension from my dialplan. I have tried this on both ends of the conversation and also tried hitting # again after entering the extension and still no luck. One end of the conversation is a SNOM 320, the other is an outside line. ABE entitles you (or should) to support directly from digium, so if you dont get help here you should take them up on that :) However with that said, have you looked in features.conf? It may be disabled for the transfer. To allow the # to be transmitted to remote systems (like banks and other institutions that require the # key) often transfering is mapped to #1 for blind. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] transfers from Polycom 501 involving Sipura 300 and asterisk 1.2
When transferring a call that came in on the Sipura and picked up by a Polycom 501 (sip 1.52), then transferred to another polycom using the transfer button on the polycom (havn't tried with the blind transfer from the polycom phone), then as soon as the transfer is complete (after pressing transfer again on the polycom) then the caller on the Sipura side can hear the new polycom caller, but the polycom cannot hear the sipura caller. This is all on a flat network, no nat, no gateways, between any of the points. If I change canreinvite=no for the sipura then everyting works fine. I'm assuming this is a bug in 1.2, but before I jump to conclusions I would like to know if anyone else has seen this? I did not yet have a chance to capture the output, but will do so if needed. Thank You ___ --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] Transfers on PRI connected channel banks and legacy PBX's
Hi, Were using our legacy PBX as a channel bank with asterisk sitting between the pbx and our telco provider spliced by a TE410P. If it were a straight analog FXS card then wed use a hook flash to break into asterisk for transfers etc, does anybody know what the equivalent is for the PRI zaptel support? Regards Steve Steve Hanselman Brendata (UK) Ltd Tel: +44 (0)1268 466111 Fax: +44 (0)870 1387283 Mob: +44 (0)7973 750993 The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ 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] # Transfers
I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom SIP phones, running 1.4.1. Too many of our transfers using the Transfer end up with zombie channels after a REFER. As such, I implemented # transfers, and all is well. Sort of. I have a reproducible issue. Take a call from a queue. Press #, and it'll transfer just fine. Now, take a call from the queue. Put them on hold for a couple seconds. Pick them back up and press #. They hear a beautiful, short, DTMF tone, nothing more. Is this a bug, or did I miss something in the configurations? Has anyone else had this problem? As far as the transfers, I found a message at http://lists.digium.com/pipermail/asterisk-users/2004-September/062080.h tml but there were no more messages in that thread. The other zombie channel transfer questions didn't seem to fit the problem, but I may be wrong. Any suggestions would be greatly appreciated. Thanks, David ___ 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] # Transfers
David, Check out bug number 4375. Does this relate? 4375 is plaquing us like mad and if I can find more people that get this too then it might move up in priority. -Matthew David Gomillion wrote: I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom SIP phones, running 1.4.1. Too many of our transfers using the Transfer end up with zombie channels after a REFER. As such, I implemented # transfers, and all is well. Sort of. I have a reproducible issue. Take a call from a queue. Press #, and it'll transfer just fine. Now, take a call from the queue. Put them on hold for a couple seconds. Pick them back up and press #. They hear a beautiful, short, DTMF tone, nothing more. Is this a bug, or did I miss something in the configurations? Has anyone else had this problem? As far as the transfers, I found a message at http://lists.digium.com/pipermail/asterisk-users/2004-September/062080.h tml but there were no more messages in that thread. The other zombie channel transfer questions didn't seem to fit the problem, but I may be wrong. Any suggestions would be greatly appreciated. Thanks, David ___ 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] Transfers tend to fail after upgrade to 1.0.7
Hi We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected to old PBX, and some SIP phones, used by a callcenter with queues. Almost all calls are incoming (through E1 line), answered by some callcenter operator (using SIP phones, call assigned by queue app), and in some cases, are transferred to some other extension on the old PBX or other SIP. We had problems with Music on Hold (on the queue) and with transfers on version 1.0.3. We now upgraded to 1.0.7 and the MoH problem is gone, but we still have some transfer problems. What happens is that sometimes when one callcenter op (SIP client) does a transfer to another SIP or an extension that is mapped to a FXO line (old PBX), we get a half-call: the caller hears the called station, but the called station (the one the call is transferred to) does not here the caller. As we need attended transfer, the calls are made from the SIP phone (Xten), using the transfer button (not blind transfers). Don't really know how to debug this. Is there a log I can see that can help me pinpoint the problem?. On that log, what should we be looking for? I'm used to debug this kind of problems in general, but are not familiar with SIP protocol nor Asterisk debugging. We tried to change SIP phones, but its the same. Note that it happens with calls that have one end on the E1 and the other to FXO, both local to Asterisk (joined by a SIP phone), so it does not seems to be a codec problem. Thanks for any advice, Pablo PD: I sent this mail some days ago, but didn't see it on the list. My apologies if it is a dupe. ___ 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] Transfers fails, even after upgrade to 1.0.7
Hi We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected to old PBX, and some SIP phones, used by a callcenter with queues. Almost all calls are incoming (through E1 line), answered by some callcenter operator (using SIP phones, call assigned by queue app), and in some cases, are transferred to some other extension on the old PBX or other SIP. We had problems with Music on Hold (on the queue) and with transfers on version 1.0.3. We now upgraded to 1.0.7 and the MoH problem is gone, but we still have some transfer problems. What happens is that sometimes when one callcenter op (SIP client) does a transfer to another SIP or an extension that is mapped to a FXO line (old PBX), we get a half-call: the caller hears the called station, but the called station (the one the call is transferred to) does not here the caller. As we need attended transfer, the calls are made from the SIP phone (Xten), using the transfer button (not blind transfers). Don't really know how to debug this. Is there a log I can see that can help me pinpoint the problem?. On that log, what should we be looking for? I'm used to debug this kind of problems in general, but are not familiar with SIP protocol nor Asterisk debugging. We tried to change SIP phones, but its the same. Note that it happens with calls that have one end on the E1 and the other to FXO, both local to Asterisk (joined by a SIP phone), so it does not seems to be a codec problem. Thanks for any advice. ___ 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] Transfers to engaged extensions
Hi, I'm using zaptel with three way calling and call transfers with a hookflash. If I try transfering a call to an extension that is engaged I get an engaged tone. This is fine, but how do I get back to the caller? If I hookflash again I seem to put on a three-way call and the caller can hear the beeping. I can press hookflash again but I'd prefer the caller to hear only the hold music and then me speaking. Is this intentional or am I doing something wrong? Robie. ___ 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] # Transfers.
What needs to be done to make this work? For me, this would be the only time we'd really use attended transfers, on the way from an agent to either another agent, or a member of staff. At the moment we have to make all transfers from agents (i.e. queue calls) via blind transfer. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Kou Sent: 21 January 2005 02:02 To: Asterisk List; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] # Transfers. No, it's doesn't work. Asterisk List on 2005/1/21 01:48 wrote: I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it? Best regards, --JJL44 On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote: Does this work with app_queue/chan_agent? Cheers, Ben -- Jim Kou IT Engineer Malico Inc. Site: http://www.malico.com.tw No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel: +886-3-472-8155#2218Fax: +886-3-472-5979 __ ______ ___ _ _ _ ___ ( \/ ) /__\ ( ) (_ _)/ __)( _ ) (_ _)( \( )/ __) )( /(__)\ )(__ _)(_( (__ )(_)(_)(_ ) (( (__ (_/\/\_)(__)(__)()()\___)(_) ()(_)\_)\___)() ___ 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] # Transfers.
Thanks to Bruce for adding this stuff on attended transfers to the WIKI pages. I've been trying to get my head round this for a couple of days. Unfortunately I'm still having a bit of trouble. I have the latest CVS-HEAD, just downloaded and compiled. Added the bit for attended transfer into the Features.conf, and reloaded. However my phones just seem to ignore this. Do I need to change any other configs? Thanks Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] # Transfers.
Am Mittwoch, 19. Januar 2005 19:18 schrieb Asterisk List: The current CVS HEAD version already has ## transfer built-in. See the included configs/features.conf.sample file. You can define your own transfer key sequence. There is also an attended transfer feature. What is an attended transfer? :) -- Robert Spielmann - TAL.DE Klaus Internet Service GmbH [EMAIL PROTECTED] Robertstr. 6 * D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 * Fax +49 (0) 202 / 495-399 ___ 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] # Transfers.
Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ 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] # Transfers.
Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 20 Jan 2005, Asterisk List wrote: Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-20%5Ce78d2d987a5e46cca50a486612386c7fC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] # Transfers.
features.conf bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Thursday, January 20, 2005 11:05 AM To: Asterisk List Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 20 Jan 2005, Asterisk List wrote: Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005- 01-20%5Ce78d2d987a5e46cca50a486612386c7fC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] # Transfers.
I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] # Transfers.
Does this work with app_queue/chan_agent? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] # Transfers.
I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it? Best regards, --JJL44 On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote: Does this work with app_queue/chan_agent? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 -- --JJL44 ___ 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] # Transfers.
No, it's doesn't work. Asterisk List on 2005/1/21 01:48 wrote: I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it? Best regards, --JJL44 On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote: Does this work with app_queue/chan_agent? Cheers, Ben -- Jim Kou IT Engineer Malico Inc. Site: http://www.malico.com.tw No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel: +886-3-472-8155#2218Fax: +886-3-472-5979 __ ______ ___ _ _ _ ___ ( \/ ) /__\ ( ) (_ _)/ __)( _ ) (_ _)( \( )/ __) )( /(__)\ )(__ _)(_( (__ )(_)(_)(_ ) (( (__ (_/\/\_)(__)(__)()()\___)(_) ()(_)\_)\___)() ___ 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] # Transfers.
The current CVS HEAD version already has ## transfer built-in. See
the included configs/features.conf.sample file. You can define your
own transfer key sequence. There is also an attended transfer
feature.
features.conf file:
[featuremap]
blindxfer = ##
atxfer = **
This worked very well for me.
On Wed, 19 Jan 2005 00:32:15 -0500, Ronald Hartmann
[EMAIL PROTECTED] wrote:
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not Enter my PIN followed by Pound
Likewise if I turn off the ability to transfer when initiating a call,
my bank pin works great, however I loose that ability to call park the
person I called
So I can pass the call to someone else in the office.
Conf file for park
[parkedcalls]
exten = 70,1,Answer
exten = 70,2,SetMusicOnHold(default)
exten =
70,3,ParkAndAnnounce(PARKED,10,SIP/${DIALED-EXTEN}|ext-local,${DIALED-EX
TEN},1)
exten = 70,4,Hangup
exten = _7X[1-9],1,ParkedCall(${EXTEN})
So I could adopt the doublehash patch. but it does not seam to be
something to make the CVS. therefore I have to patch patch patch
repeatedly.
What is everyone else using.
If pound pound is not something Mark the asterisk God does not wish to
add to CVS, would something like the following work
# would work as normal Conduct a pound transfer
However *#, or *## would send a Pound in DTMF to the called party.
This way it will keep the Pound Transfer in tact.
Anyways I ramble, I am anxious to see how others much brighter than I
have solved this issue.
ron
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[Asterisk-Users] # Transfers.
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not Enter my PIN followed by Pound
Likewise if I turn off the ability to transfer when initiating a call,
my bank pin works great, however I loose that ability to call park the
person I called
So I can pass the call to someone else in the office.
Conf file for park
[parkedcalls]
exten = 70,1,Answer
exten = 70,2,SetMusicOnHold(default)
exten =
70,3,ParkAndAnnounce(PARKED,10,SIP/${DIALED-EXTEN}|ext-local,${DIALED-EX
TEN},1)
exten = 70,4,Hangup
exten = _7X[1-9],1,ParkedCall(${EXTEN})
So I could adopt the doublehash patch. but it does not seam to be
something to make the CVS. therefore I have to patch patch patch
repeatedly.
What is everyone else using.
If pound pound is not something Mark the asterisk God does not wish to
add to CVS, would something like the following work
# would work as normal Conduct a pound transfer
However *#, or *## would send a Pound in DTMF to the called party.
This way it will keep the Pound Transfer in tact.
Anyways I ramble, I am anxious to see how others much brighter than I
have solved this issue.
ron
oledata.mso
Description: Binary data
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Re: [Asterisk-Users] # Transfers.
Theres a couple of ways -
Check to see if your bank really requires you to press pound. Mine says to
press it, but all pins are fixed length so it may time out after a second or
two.
Alternatively put a regex in your dialplan to recognise the phone banking
and bill payment numbers and call the dial command without transfer. For
example here in Australia most banks, bill payments and other information
services start with 13 so I match the 13 prefix in the dial plan and dial
without the transfer.
Craig
- Original Message -
From: Ronald Hartmann [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 19, 2005 1:32 PM
Subject: [Asterisk-Users] # Transfers.
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not Enter my PIN followed by Pound
Likewise if I turn off the ability to transfer when initiating a call,
my bank pin works great, however I loose that ability to call park the
person I called
So I can pass the call to someone else in the office.
Conf file for park
[parkedcalls]
exten = 70,1,Answer
exten = 70,2,SetMusicOnHold(default)
exten =
70,3,ParkAndAnnounce(PARKED,10,SIP/${DIALED-EXTEN}|ext-local,${DIALED-EX
TEN},1)
exten = 70,4,Hangup
exten = _7X[1-9],1,ParkedCall(${EXTEN})
So I could adopt the doublehash patch. but it does not seam to be
something to make the CVS. therefore I have to patch patch patch
repeatedly.
What is everyone else using.
If pound pound is not something Mark the asterisk God does not wish to
add to CVS, would something like the following work
# would work as normal Conduct a pound transfer
However *#, or *## would send a Pound in DTMF to the called party.
This way it will keep the Pound Transfer in tact.
Anyways I ramble, I am anxious to see how others much brighter than I
have solved this issue.
ron
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[Asterisk-Users] transfers with zap channel
Ok, I've seen discussion before on doing transfers
(attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying (unsuccessfully) to
do transfers with a zap channel (analog phone attached to TDM400). I have
no idea what I'm missing. My current understanding is that I need to have
transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, flash hook
again and we're linked (attended). Then if I hang up the call will
be transfered.
However, when I try to do this things don't
work. Here's what I do:
- connection is made between Zap/3 (analog phone)
and Zap/1 (outside line).
- flash hook to get dialtone (I do get
dialtone)
- attempt to transfer to extension 7007 - I dial
7007
- after dialing the 2nd zero, and before dialing
the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is
hung up (I get a busy signal). Zap/1 gets parked.
Here's what the log shows:
-- Zap/1-1 answered
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Started three way call on channel
1 -- Started music on hold, class 'default', on
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Starting simple switch on
'Zap/1-2' -- Started music on hold, class 'default', on
Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to dostuff,7001,1
in 45 seconds -- Added extension '701' priority 1 to
parkedcalls -- Playing 'digits/7' (language
'en') -- Hungup 'Zap/1-1' == Spawn extension
(dostuff, 7001, 1) exited non-zero on
'Parked/Zap/3-1ZOMBIE' -- Stopped music on hold on
Parked/Zap/3-1ZOMBIE -- Playing 'digits/0'
(language 'en') -- Playing 'digits/1' (language
'en') -- Parking call to 'Zap/1-2'
-- Hungup 'Zap/1-2' -- Stopped music on hold on
Zap/3-1 == Zap/3-1 got tired of being parked --
Hungup 'Zap/3-1'
I'm not sure what I'm missing. Apparently
something to do with parked calls, so I must be misunderstanding how do to the
call transfer.
I've also tried enabling Asterisk transfer on the
channel (exten = 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this allows
one to hit the pound (#) to start a transfer. Yet pound does
nothing. Is it fair to assume that the tT only works on SIP channels, or
am I missing something else.
Any help is much appreciated
Paul
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Re: [Asterisk-Users] transfers with zap channel
How long between getting parked is the orginal call
dropping?
Depending on your dialplan, yes dialing 700x will
almost immediately send the call to call parking. (IMHO, poor extension planning
can also cause this.)
I don't use the t or T optionsPERIOD.
IMHO, you just lose the ability to use the # key and confused the heck out of my
users. Took it out and use the flash method only in my dial plan.
Dial 700, park the call. Dial the other extension, tell them to pick up
701. Or use meetme for conference calling?
I know I need to play with three way calling here
also.
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 6:12
PM
Subject: [Asterisk-Users] transfers with
zap channel
Ok, I've seen discussion before on doing
transfers (attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying (unsuccessfully)
to do transfers with a zap channel (analog phone attached to TDM400). I
have no idea what I'm missing. My current understanding is that I need
to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd number,
flash hook again and we're linked (attended). Then if I hang up
the call will be transfered.
However, when I try to do this things don't
work. Here's what I do:
- connection is made between Zap/3 (analog phone)
and Zap/1 (outside line).
- flash hook to get dialtone (I do get
dialtone)
- attempt to transfer to extension 7007 - I dial
7007
- after dialing the 2nd zero, and before dialing
the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is
hung up (I get a busy signal). Zap/1 gets parked.
Here's what the log shows:
-- Zap/1-1 answered
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Started three way call on channel
1 -- Started music on hold, class 'default', on
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Starting simple switch on
'Zap/1-2' -- Started music on hold, class 'default', on
Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to
dostuff,7001,1 in 45 seconds -- Added extension '701'
priority 1 to parkedcalls -- Playing 'digits/7'
(language 'en') -- Hungup 'Zap/1-1' == Spawn
extension (dostuff, 7001, 1) exited non-zero on
'Parked/Zap/3-1ZOMBIE' -- Stopped music on hold
on Parked/Zap/3-1ZOMBIE -- Playing 'digits/0'
(language 'en') -- Playing 'digits/1' (language
'en') -- Parking call to 'Zap/1-2'
-- Hungup 'Zap/1-2' -- Stopped music on hold on
Zap/3-1 == Zap/3-1 got tired of being parked
-- Hungup 'Zap/3-1'
I'm not sure what I'm missing. Apparently
something to do with parked calls, so I must be misunderstanding how do to the
call transfer.
I've also tried enabling Asterisk transfer on the
channel (exten = 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this
allows one to hit the pound (#) to start a transfer. Yet pound does
nothing. Is it fair to assume that the tT only works on SIP channels, or
am I missing something else.
Any help is much appreciated
Paul
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Re: [Asterisk-Users] transfers with zap channel
The outside line isn't actually being dropped - the
outside line hanging up is me hanging up the outside line after finding that my
transfer failed.
I must be not understanding how the flash-hook
works then. My understanding was that when I flash-hook and get a second
dialtone I should be able to dial the extention I want to reach (7007 is another
extension, via SIP). Normally, if I pick up the analog phone and dial 7007
it rings the extention fine. Apparently, though, when you get that
second dialtone, it has different rules? I haven't been able to find
documentation on this, can it be found anywhere? For example, why does
dialing 700 park the call? I haven't found anything on this...
*shrug*...
Paul
- Original Message -
From:
Lyle
Giese
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 7:22
PM
Subject: Re: [Asterisk-Users] transfers
with zap channel
How long between getting parked is the orginal
call dropping?
Depending on your dialplan, yes dialing 700x will
almost immediately send the call to call parking. (IMHO, poor extension
planning can also cause this.)
I don't use the t or T
optionsPERIOD. IMHO, you just lose the ability to use the # key
and confused the heck out of my users. Took it out and use the flash
method only in my dial plan. Dial 700, park the call. Dial the
other extension, tell them to pick up 701. Or use meetme for conference
calling?
I know I need to play with three way calling here
also.
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 6:12
PM
Subject: [Asterisk-Users] transfers
with zap channel
Ok, I've seen discussion before on doing
transfers (attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying
(unsuccessfully) to do transfers with a zap channel (analog phone attached
to TDM400). I have no idea what I'm missing. My current
understanding is that I need to have transfer=yes in zapata.conf, do a flash
hook, dial the 2nd number, flash hook again and we're linked
(attended). Then if I hang up the call will be
transfered.
However, when I try to do this things don't
work. Here's what I do:
- connection is made between Zap/3 (analog
phone) and Zap/1 (outside line).
- flash hook to get dialtone (I do get
dialtone)
- attempt to transfer to extension 7007 - I
dial 7007
- after dialing the 2nd zero, and before
dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and
then Zap/3 is hung up (I get a busy signal). Zap/1 gets
parked.
Here's what the log shows:
-- Zap/1-1 answered
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Started three way call on channel
1 -- Started music on hold, class 'default', on
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Starting simple switch on
'Zap/1-2' -- Started music on hold, class 'default',
on Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to
dostuff,7001,1 in 45 seconds -- Added extension '701'
priority 1 to parkedcalls -- Playing 'digits/7'
(language 'en') -- Hungup 'Zap/1-1' == Spawn
extension (dostuff, 7001, 1) exited non-zero on
'Parked/Zap/3-1ZOMBIE' -- Stopped music on
hold on Parked/Zap/3-1ZOMBIE -- Playing
'digits/0' (language 'en') -- Playing 'digits/1'
(language 'en') -- Parking call to
'Zap/1-2' -- Hungup 'Zap/1-2' --
Stopped music on hold on Zap/3-1 == Zap/3-1 got tired of being
parked -- Hungup 'Zap/3-1'
I'm not sure what I'm missing. Apparently
something to do with parked calls, so I must be misunderstanding how do to
the call transfer.
I've also tried enabling Asterisk transfer on
the channel (exten = 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this
allows one to hit the pound (#) to start a transfer. Yet pound does
nothing. Is it fair to assume that the tT only works on SIP channels,
or am I missing something else.
Any help is much appreciated
Paul
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Re: [Asterisk-Users] transfers with zap channel
Have you looked at features.conf?
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 8:53
PM
Subject: Re: [Asterisk-Users] transfers
with zap channel
The outside line isn't actually being dropped -
the outside line hanging up is me hanging up the outside line after finding
that my transfer failed.
I must be not understanding how the flash-hook
works then. My understanding was that when I flash-hook and get a second
dialtone I should be able to dial the extention I want to reach (7007 is
another extension, via SIP). Normally, if I pick up the analog phone and
dial 7007 it rings the extention fine. Apparently, though, when
you get that second dialtone, it has different rules? I haven't
been able to find documentation on this, can it be found anywhere? For
example, why does dialing 700 park the call? I haven't found anything on
this... *shrug*...
Paul
- Original Message -
From:
Lyle
Giese
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 7:22
PM
Subject: Re: [Asterisk-Users] transfers
with zap channel
How long between getting parked is the orginal
call dropping?
Depending on your dialplan, yes dialing 700x
will almost immediately send the call to call parking. (IMHO, poor extension
planning can also cause this.)
I don't use the t or T
optionsPERIOD. IMHO, you just lose the ability to use the #
key and confused the heck out of my users. Took it out and use the
flash method only in my dial plan. Dial 700, park the call. Dial
the other extension, tell them to pick up 701. Or use meetme for
conference calling?
I know I need to play with three way calling
here also.
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List
- Non-Commercial Discussion
Sent: Monday, January 17, 2005 6:12
PM
Subject: [Asterisk-Users] transfers
with zap channel
Ok, I've seen discussion before on doing
transfers (attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying
(unsuccessfully) to do transfers with a zap channel (analog phone attached
to TDM400). I have no idea what I'm missing. My current
understanding is that I need to have transfer=yes in zapata.conf, do a
flash hook, dial the 2nd number, flash hook again and we're linked
(attended). Then if I hang up the call will be
transfered.
However, when I try to do this things don't
work. Here's what I do:
- connection is made between Zap/3 (analog
phone) and Zap/1 (outside line).
- flash hook to get dialtone (I do get
dialtone)
- attempt to transfer to extension 7007 - I
dial 7007
- after dialing the 2nd zero, and before
dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and
then Zap/3 is hung up (I get a busy signal). Zap/1 gets
parked.
Here's what the log shows:
-- Zap/1-1 answered
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Started three way call on channel
1 -- Started music on hold, class 'default', on
Zap/3-1 -- Attempting native bridge of Zap/3-1 and
Zap/1-1 -- Starting simple switch on
'Zap/1-2' -- Started music on hold, class 'default',
on Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to
dostuff,7001,1 in 45 seconds -- Added extension
'701' priority 1 to parkedcalls -- Playing
'digits/7' (language 'en') -- Hungup
'Zap/1-1' == Spawn extension (dostuff, 7001, 1) exited non-zero
on 'Parked/Zap/3-1ZOMBIE' -- Stopped music
on hold on Parked/Zap/3-1ZOMBIE -- Playing
'digits/0' (language 'en') -- Playing 'digits/1'
(language 'en') -- Parking call to
'Zap/1-2' -- Hungup 'Zap/1-2'
-- Stopped music on hold on Zap/3-1 == Zap/3-1 got tired of
being parked -- Hungup 'Zap/3-1'
I'm not sure what I'm missing.
Apparently something to do with parked calls, so I must be
misunderstanding how do to the call transfer.
I've also tried enabling Asterisk transfer on
the channel (exten = 7010,1,Dial(${CORDLESS},20,tT)).
My understanding of this method is that this
allows one to hit the pound (#) to start a transfer. Yet pound does
nothing. Is it fair to assume that the tT only works on SIP
channels, or am I missing something else.
Any help is much appreciated
Paul
Re: [Asterisk-Users] transfers with zap channel
Ah, suddenly everything becomes clear. I've never looked in features.conf before. I now understand that 700 is supposed to intitiate the call park, and it's taking precidence over the extension I was trying to dial of 7007. I've changed the call parking extension and now I can do regular attended and unattended transfers without having to park the call... (note to anyone else changing features.conf, you have to 'restart' asterisk, a 'reload' won't do). thanks a bunch for the help, guys... Paul - Original Message - From: Lyle Giese To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 8:20 PM Subject: Re: [Asterisk-Users] transfers with zap channel Have you looked at features.conf? Lyle - Original Message - From: Paul Fielding To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 8:53 PM Subject: Re: [Asterisk-Users] transfers with zap channel The outside line isn't actually being dropped - the outside line hanging up is me hanging up the outside line after finding that my transfer failed. I must be not understanding how the flash-hook works then. My understanding was that when I flash-hook and get a second dialtone I should be able to dial the extention I want to reach (7007 is another extension, via SIP). Normally, if I pick up the analog phone and dial 7007 it rings the extention fine. Apparently, though, when you get that second dialtone, it has different rules? I haven't been able to find documentation on this, can it be found anywhere? For example, why does dialing 700 park the call? I haven't found anything on this... *shrug*... Paul - Original Message - From: Lyle Giese To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 7:22 PM Subject: Re: [Asterisk-Users] transfers with zap channel How long between getting parked is the orginal call dropping? Depending on your dialplan, yes dialing 700x will almost immediately send the call to call parking. (IMHO, poor extension planning can also cause this.) I don't use the t or T optionsPERIOD. IMHO, you just lose the ability to use the # key and confused the heck out of my users. Took it out and use the flash method only in my dial plan. Dial 700, park the call. Dial the other extension, tell them to pick up 701. Or use meetme for conference calling? I know I need to play with three way calling here also. Lyle - Original Message - From: Paul Fielding To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 6:12 PM Subject: [Asterisk-Users] transfers with zap channel Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, flash hook again and we're linked (attended). Then if I hang up the call will be transfered. However, when I try to do this things don't work. Here's what I do: - connection is made between Zap/3 (analog phone) and Zap/1 (outside line). - flash hook to get dialtone (I do get dialtone) - attempt to transfer to extension 7007 - I dial 7007 - after dialing the 2nd zero, and before dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is hung up (I get a busy signal). Zap/1 gets parked. Here's what the log shows: -- Zap/1-1 answered Zap/3-1 -- Attempting native bridge of Zap/3-1 and Zap/1-1 -- Started three way call on channel 1 -- Started music on hold, class 'default', on Zap/3-1 -- Attempting native bridge of Zap/3-1 and Zap/1-1 -- Starting simple switch on 'Zap/1-2' -- Started music on hold, class 'default', on Zap/3-1 == Parked Zap/3-1 on 701. Will timeout back to dostuff,7001,1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Hungup 'Zap/1-1' == Spawn extension (dostuff, 7001, 1) exited non-zero on 'Parked/Zap/3
[Asterisk-Users] Transfers- Please Help ASAP
Can someone please help with this. After an outside caller has been parked, they inherit our abilitites to transfer. I have played with all the different combinations of T and t, but nothing seems to work. I found a way to get my Sipura to work with a flash transfer. So right now I am stuck. Is there any way to block callers that have been parked from being able to park us. The only time that this happens is after the call has been parked. Plase help! Im at a loss and people here want answers. ___ 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] Transfers (sip or asterisk #' base) broken in certain scenario
I've got an interesting scenario where transfers while getting an invite seem to break. Here is the scenario: You have a receptionist who has a 6 line phone (in this case, a polycom ip600 - also tested with a Cisco 7960) the receptionist has all six lines available for use (in the case of the cisco I tried registering all lines as one number as well as registering multiple lines and having the dialplan do roll-over). The receptionist receives a call and begins to transfer the call and in the middle of transferring the call, another call is received. This is what happens: If the receptionist is using the cisco/polycom soft button labelled transfer, the transfer goes through, however, the receptionist never knows another call was coming in. It went straight to the 'busy' priority (+101 in the case of a single registered extension or +101, +101, +101, +101... all the way through a roll-over dialplan straight to busy handling even though 5 of the 6 lines were available with no active calls) In the case of using # to effect a transfer, the receptionist hits pound and begins entering the phone number to transfer to and a call comes in. Immediately the receptionist is send to the 'i' extension while doing the transfer, and the new call is presented (rings and LCD screen shows information). In some cases depending on the timing of the new call (if it's received after pressing # but before entering an extension to transfer to) you can get the call back, place it on hold and take the new call, however if you are in the process of entering the number to transfer when the new call comes in, then the original call is immediately acted on. In other words, if I was typing 2004 and had entered 20 when the new call came in, asterisk grabs the 20 and tries to transfer the call to it. No matter what happens, the call is lost to the receptionist, unable to get it back, even if there is a valid 'i' handler. Is there anyone out there who has a busy enough system to have seen this as well? If so, how have you dealt with it? The only solution I can think of is to place all inbound calls into a queue, then pass them to the receptionist as the only agent (permanent agent) of the queue. Then set limits on the number of calls the receptionist is allowed to have incoming 1, outgoing 1 so the queue won't ring the receptionists phone unless there is no active sessions. Relevant info below Asterisk CVS-D2004.07.03.19.00.00-07/05/04-14:41:51 Polycom IP500 sip.ld version 1.2.0.0318 Cisco 7960 SIP image 6.0 sip.conf entries are simple and are not a factor in the problem - items that may be important from sip.conf are: canreinvite=no ; want asterisk in the media stream for features type=friend ; haven't tried this by creating a user and peer for each handset (yuck) extensions.conf entries are either one of the following (tested against both) exten = _,1,Dial(SIP/1000,30,t) exten = _,2,Voicemail(u1000) exten = _,3,Hangup exten = _,102,Voicemail(b1000) exten = _,103,Hangup exten = _,1,Dial(SIP/1000,30,t) exten = _,2,Voicemail(u1000) exten = _,3,Hangup exten = _,102,Dial(SIP/1001,30,t) exten = _,103,Voicemail(u1000) exten = _,104,Hangup exten = _,203,Dial(SIP/1002,30,t) ... exten = _,506,Dial(SIP/1005,30,t) exten = _,507,Voicemail(u1000) exten = _,508,Hangup exten = _,607,Voicemail(b1000) exten = _,608,Hangup -Chris ___ 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] Transfers on the Cisco 7960
On Thu, 2003-07-10 at 10:26, Kim C. Callis wrote: I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis The transfer button is for Attended transfers Which is when you call the person, announce there is a call for them, and then transfer it to them. On the opposite side of the display as the Transfer soft button is a BldXfer. If you hit this button and dial a number it will do a unannounced transfer which is what it sounds like you're looking for. To use the Attended Transfer feature, hit the transfer button, dial the other extension, announce the call to the person on the other extension, then hit the transfer soft button again to complete the transfer. -- Ryan Butler [EMAIL PROTECTED] ADI Internet Solutions ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfers on the Cisco 7960
We support both blind and supervised transfers on the Cisco. Mark On Thu, 10 Jul 2003, Kim C. Callis wrote: I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfers on the Cisco 7960
On Thursday 10 July 2003 03:24 pm, Mark Spencer wrote: We support both blind and supervised transfers on the Cisco. Mark On Thu, 10 Jul 2003, Kim C. Callis wrote: I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis So what SIP phones are the most compatible and have the most features with asterix ??? Is there a list somewhere ?? Regards...Martin -- modem, adj.: Up-to-date, new-fangled, as in Thoroughly Modem Millie. An unfortunate byproduct of kerning. [That's sic!] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
