[asterisk-users] atxfer fails to read data
Hi, We are having a problem that is preventing users from using *2 to manage an attended transfer. After dialling *2, asterisk places the call on hold, but you can only dial one digit before it times out, and the cli says:- [2011-06-29 18:33:55] WARNING[29877]: res_features.c:938 builtin_atxfer: Did not read data. There is already an issue in JIRA: https://issues.asterisk.org/jira/browse/ASTERISK-16927 And loads of people on forums with this problem, but no one has an answer. Have anyone made a workaround for this? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Atxfer Command
Hi, We are testing new Asterisk 1.6.0.1 because we would like to use the Attended Transfer feature and we are trying to use the new action Atxfer developed for AMI. As far as we know, it is suposed to be in this release as it can be read in Digium's changelog /New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/ But, when we try to see information in the CLI console, this command doesn't exist, neither AtxferAction works, we got a message saying that this command is unknown Are we missing something? Thanks in advance David ___ -- 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] Atxfer Command
David Monteagudo Sanz wrote: Hi, We are testing new Asterisk 1.6.0.1 because we would like to use the Attended Transfer feature and we are trying to use the new action Atxfer developed for AMI. As far as we know, it is suposed to be in this release as it can be read in Digium's changelog /New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/ But, when we try to see information in the CLI console, this command doesn't exist, neither AtxferAction works, we got a message saying that this command is unknown Are we missing something? Thanks in advance David The AMI atxfer command is not in 1.6.0 and will be in no releases of 1.6.0. It is in 1.6.1, for which a beta is currently available. Mark Michelson ___ -- 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] atxfer attended transfer feature
--- Don Pobanz [EMAIL PROTECTED] wrote: I would like to know if atxfer is supported This was a little confusing for me also. A week or so ago, someone pointed out that you need to include featuremap in your extensions.conf Thanks Don I'll try that. It surprises me that such an important feature is vaguely documented in *. Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] atxfer attended transfer feature
I would like to know if atxfer is supported somehow because there seems to be little documentation for this feature. I know most people expect a good SIP/IAX phone to do the job but I think it's nice to be able to do attended trasnfers with a simple ATA-connected analog phone. I have Asterisk 1.2/Freepbx and features.conf has a line regarding atxfer and I set it to *2 (Default). While # works fine (blind transfers), *2 doesn't. Can someone please tell me if I should just forget using the atxfer feature or suggest me a link to some info other than what's on voip-info. Also, has atxfer improved in Asterisk 1.4 ? Thanks Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --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] atxfer attended transfer feature
I would like to know if atxfer is supported somehow because there seems to be little documentation for this feature. ... I have Asterisk 1.2/Freepbx and features.conf has a line regarding atxfer and I set it to *2 (Default). While # works fine (blind transfers), *2 doesn't. This was a little confusing for me also. A week or so ago, someone pointed out that you need to include featuremap in your extensions.conf in order to enable attended transfer, even though it appears in features.conf. I guess blind transfer works regardless of if you include featuremap or not. Don Pobanz ___ --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] atxfer attended transfer feature
Vieri wrote: --- Don Pobanz [EMAIL PROTECTED] wrote: I would like to know if atxfer is supported This was a little confusing for me also. A week or so ago, someone pointed out that you need to include featuremap in your extensions.conf Thanks Don I'll try that. It surprises me that such an important feature is vaguely documented in *. It is not all that important of a feature as most VoIP devices (including ATAs) have this feature built in. ___ --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] atxfer not working
Hi, I cannot get attended working on my Asterisk 1.2.9.1 during an inbound call via an ISDN card to a Snom SIP phone. The called party is not able to transfer even if : 1 - atxfer is enabled (set to *7) in in features.conf 2 - the dial option is set to value 't' 3 - I see * and then 7 on Asterisk CLI when debug is set to DTMF Asterisk gets the right sequence from Snom phone (CLI does not lie) but for some reason Asterisk is not transferring the call while the caller keeps on speaking with the called party. Is there anybody who knows what is going wrong? TIA Giorgio Incantalupo ___ --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] ATXFER
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Re: [Asterisk-Users] ATXFER
On Friday 12 May 2006 21:00, Josué Conti wrote: Eric, but I to continue using version 1.0.9 of asterisk, would have some solution? Yes, find a consultant to try and backport this feature to the version of Asterisk you desire. -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] ATXFER
Andrew, thank´s for the attention, but necessary to decide this my problem. You he could help me? Regards Josué 2006/5/14, Andrew Kohlsmith [EMAIL PROTECTED]: On Friday 12 May 2006 21:00, Josué Conti wrote: Eric, but I to continue using version 1.0.9 of asterisk, would have some solution?Yes, find a consultant to try and backport this feature to the version ofAsterisk you desire.-A.___--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] ATXFER
How many times do u need to repeat it?. You could change this info via web list manager. I think you need to read how to do that before sending 20 emails with same subject. [EMAIL PROTECTED] escribió: Please change the email address of [EMAIL PROTECTED] to [EMAIL PROTECTED] 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 ___ --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] ATXFER
I think it's most likely that it's a mail loop caused by a brain dead 'change of address' script. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Saturday, 13 May 2006 17:00 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ATXFER How many times do u need to repeat it?. You could change this info via web list manager. I think you need to read how to do that before sending 20 emails with same subject. [EMAIL PROTECTED] escribió: Please change the email address of [EMAIL PROTECTED] to [EMAIL PROTECTED] 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 ___ --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] ATXFER
Eric, thank you very much. But It could help in this case me? Regards Josué 2006/5/12, Eric ManxPower Wieling [EMAIL PROTECTED]: Josué Conti wrote: Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk and it is really all good with this version, only this case of atxfer that it does not function. The function DYNAMIC_FEATURE = to atxfer in my [ globals ] of extensions.conf functions in version 1.0.9? It could help in this case me? Best Regards1.0.x does not support this.--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___ --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] ATXFER
If you want dynamic features you must upgrade to Asterisk 1.2.x Josué Conti wrote: Eric, thank you very much. But It could help in this case me? Regards Josué 2006/5/12, Eric ManxPower Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Josué Conti wrote: Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk and it is really all good with this version, only this case of atxfer that it does not function. The function DYNAMIC_FEATURE = to atxfer in my [ globals ] of extensions.conf functions in version 1.0.9? It could help in this case me? Best Regards 1.0.x does not support this. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] ATXFER
Eric, but I to continue using version 1.0.9 of asterisk, would have some solution? Regards Josué 2006/5/12, Eric ManxPower Wieling [EMAIL PROTECTED]: If you want dynamic features you must upgrade to Asterisk 1.2.xJosué Conti wrote: Eric, thank you very much. But It could help in this case me? Regards Josué 2006/5/12, Eric ManxPower Wieling [EMAIL PROTECTED] mailto: [EMAIL PROTECTED]: Josué Conti wrote: Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk and it is really all good with this version, only this case of atxfer that it does not function. The function DYNAMIC_FEATURE = to atxfer in my [ globals ] of extensions.conf functions in version 1.0.9? It could help in this case me? Best Regards 1.0.x does not support this.--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___ --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] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
Can someone please kill this guy's account? Isn't there a Moderator on this list? bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, May 12, 2006 10:05 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ATXFER Please change the email address of [EMAIL PROTECTED] to [EMAIL PROTECTED] 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 __ NOD32 1.1535 (20060512) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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RE: [Asterisk-Users] ATXFER
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Re: [Asterisk-Users] ATXFER
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[Asterisk-Users] ATXFER
Hi all. Iam with the following problem: I have one perfectly queue functioning, however when the agent receives a call and effects a transference the blind people, blindxfer, asterisk functions informs to transfer and the transference is ok. However, if the agent tries to effect an attended transference the ATXFER, knocks down the call. All the agents of this queue are with canreinvite=no in sip.conf and parameters t T are qualified in extensions.conf and mine features.conf is qualified [ featuremap ] with the code of atxfer. Best Regards Josue ___ --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] ATXFER
On 05/11/06 19:46 Josué Conti said the following: functions informs to transfer and the transference is ok. However, if the agent tries to effect an attended transference the ATXFER, knocks down the call. All the agents of this queue are with canreinvite=no in i'm guessing that the feature code for ATXFER in features.conf begins with a '*'. this '*' is trapped instead by chan_agent, which is a hardcoded value within chan_agent to hang up the call. that's the same symptoms you're seeing. i've submitted a patch for this, which has been committed to trunk, so the next release of asterisk should have an endcall parameter in agents.conf which allows you to turn off this feature. however, if you're using 1.2.x, and you need it now, you can apply the 1.2 related patch from http://bugs.digium.com/view.php?id=6897 apply agent-endcall.patch for 1.2.x. as noted above, this patch has already been committed to trunk. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] ATXFER
Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk and it is really all good with this version, only this case of atxfer that it does not function. The function DYNAMIC_FEATURE = to atxfer in my [ globals ] of extensions.conf functions in version 1.0.9? It could help in this case me? Best Regards Josué 2006/5/11, Dinesh Nair [EMAIL PROTECTED]: On 05/11/06 19:46 Josué Conti said the following: functions informs to transfer and the transference is ok. However, if the agent tries to effect an attended transference the ATXFER, knocks down the call. All the agents of this queue are with canreinvite=no ini'm guessing that the feature code for ATXFER in features.conf begins witha '*'. this '*' is trapped instead by chan_agent, which is a hardcodedvalue within chan_agent to hang up the call. that's the same symptomsyou're seeing.i've submitted a patch for this, which has been committed to trunk, so the next release of asterisk should have an endcall parameter in agents.confwhich allows you to turn off this feature. however, if you're using1.2.x, and you need it now, you can apply the 1.2 related patch from http://bugs.digium.com/view.php?id=6897apply agent-endcall.patch for 1.2.x. as noted above, this patch has alreadybeen committed to trunk.-- Regards, /\_/\ All dogs go to heaven.[EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+| for a in past present future; do|| for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b.|| done; done|+=+ ___--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] ATXFER
Josué Conti wrote: Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk and it is really all good with this version, only this case of atxfer that it does not function. The function DYNAMIC_FEATURE = to atxfer in my [ globals ] of extensions.conf functions in version 1.0.9? It could help in this case me? Best Regards 1.0.x does not support this. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] atxfer featuremap
Hi there i just can't find an answer on the featuremap config i want all phones to use the same method for transfering a call on all phones but i just can't get the atxfer or other functions to work on my grandsteam and sipura spa 2000 it's confusing for users with different phones to transfer a call i know you can use the transfer button but i wan't to use a code *1 not possible??? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi PF Oh, you mean the completely natural feeling put them on hold, dial PF new party, tell them you have a transfer, hit transfer? I want some of PF whatever kool-aid the person who thought that one up had. I still feel PF like I'm losing a call every time I do an attended transfer. In my opinion there should be only one transfer function, let suppose it's called by #. - You get a call - You want to transfer it - You hit # - You are presented a tone - You dial the extension you want to transfer to Now the hard part - If you hang up prior of the other party has answered you get an unattended transfer if, for any reason the other party dont answer (busy, no answer, wrong extension etc) call should be bounced back to you - If you stay on the phone and the other party answers you talk to him, introduce the call then hitting # again will switch back and forth between the call you are tranfering and the transfer party if you hang up call is trasfered to the other party if the other party hangs up you get back to the original call Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. What do you think about this flow ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ 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[4]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Hello Adam, In my opinion there should be only one transfer function, let suppose it's called by #. AG Wrong, which other phone system have you used where every time you try AG and use some IVR that says Enter your xyz number followed by the # key AG and you end up being interrupted by asterisk to transfer the call ?? Well as you can see it was an example, actually you have to decide this mapping in features.conf, so what's the point ? Let say is *# or any other sequence :) Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. AG Nope, because if there are three parties: AG A - You AG B - Outside caller 1 AG C - Outside transfer party AG When you hangup, you don't want the other two legs to stay up, AG potentially forever depending on your hangup detection etc... I know what I want! :) Why not, I'm announcing a call, then going conference, then leaving because I already did my part, why the other 2 calls have to be disconnected ... because hangup detection works bad ? What do you think about this flow ? AG Any SIP phone (decent one) should have much more intuitive/instructive AG transfer process. All I'm asking is a native function that can be used regardless of the UA, if you got such functions integrated in the phone, better yet, is up to you to choose then. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ 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: Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
On Thu, 2005-07-21 at 09:59 +0200, Alessio Focardi wrote: PF Oh, you mean the completely natural feeling put them on hold, dial PF new party, tell them you have a transfer, hit transfer? I want some of PF whatever kool-aid the person who thought that one up had. I still feel PF like I'm losing a call every time I do an attended transfer. In my opinion there should be only one transfer function, let suppose it's called by #. Wrong, which other phone system have you used where every time you try and use some IVR that says Enter your xyz number followed by the # key and you end up being interrupted by asterisk to transfer the call ?? Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. Nope, because if there are three parties: A - You B - Outside caller 1 C - Outside transfer party When you hangup, you don't want the other two legs to stay up, potentially forever depending on your hangup detection etc... What do you think about this flow ? Not only have you suggested pretty much what we have, except you've made it worse by taking away the # and * keys... If you are on a zap channel, just hook flash (or press flash/recall/whatever) and transfer the call, complete with conference option. Any SIP phone (decent one) should have much more intuitive/instructive transfer process. Regards, Adam ___ 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] ATXFER discussion, what's your opinion ?
Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. In usual pbx's normaly there is no difference between an attended call transfer and a blind one: you just hit transfer then dial the extension you want the call to be transfered. If you stay on the phone you can talk to the other party, then, when you hangup, he will get the call. If you hang immediately after the transfer sequence the call is just transfered, and if the other party is busy or does not answer the transfered call is bounced back to you again. That's how pbx's users are expecting call transfer to work, is there a way to reproduce this behavior in asterisk ? For what I can see it's not possible and you will have to select two codes, one for blind and one for attended tranfers What do you think about it ? -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ 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] ATXFER discussion, what's your opinion ?
Alessio Focardi wrote: Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. SNIP I honestly think that transfers is one thing that Asterisk should improve a LOT to be able to stand up to even the most cheapo taiwanese no-name PBXs, which support attended transfers out of the box. I've had two possible clients refuse an Asterisk installation because attended transfers were unreliable. I honestly didn't know how to explain that a feature available in PBXs for decades was not available or didn't always work. I don't know how the current HEAD is going, but so far, attended transfers weren't available in stable. Regards, Mike ___ 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[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Hello Michael, Wednesday, July 20, 2005, 11:54:40 AM, you wrote: MP Alessio Focardi wrote: Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. MP SNIP MP I honestly think that transfers is one thing that Asterisk should MP improve a LOT to be able to stand up to even the most cheapo taiwanese MP no-name PBXs, which support attended transfers out of the box. That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe we can try to write down a sort of flow chart of a new transfer function and then set up a bounty, anyone else would like to join me ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ 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: Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
On Wed, 2005-07-20 at 12:26 +0200, Alessio Focardi wrote: Wednesday, July 20, 2005, 11:54:40 AM, you wrote: MP Alessio Focardi wrote: I'm experimenting attended calls tranfers and I'm a little bit confused. MP I honestly think that transfers is one thing that Asterisk should MP improve a LOT to be able to stand up to even the most cheapo taiwanese MP no-name PBXs, which support attended transfers out of the box. That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe because most asterisk PBX's are implemented using business class softphones rather than analogue phones? Most business class SIP phones (and even grandstream phones) allow attended/non-attended transfers with asterisk-stable and/or asterisk-head... Personally I love the polycom IP600 phones, because they 'guide' the user on-screen to help them do the most simple task (like answer the phone :) PS, and yes, these are helpful features, because otherwise there really would be more support calls like the one I got today (read on for a laugh). User calls and reports that their cordless phone extension is not working (connected via TDM card). I suspect that it is the usual problem, and I need to unload/reload the wcfxs driver and restart asterisk because I haven't replaced their card with the latest revision and every few weeks it does this. Anyway, so I ask what is happening, and he says he presses the pickup button and hears 3 loud beeps, and then the phone hangs up. I ask him if the base station is plugged in, and I then hear something along the lines of Oh... ummm, yeah... thanks, cya... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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: Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe because most asterisk PBX's are implemented using business class softphones rather than analogue phones? Most business class SIP phones (and even grandstream phones) allow attended/non-attended transfers with asterisk-stable and/or asterisk-head... I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi ___ 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] ATXFER discussion, what's your opinion ?
[EMAIL PROTECTED] wrote: By the way: anyone got experience in attended trasfer with snom ? :) Works OK here, using a lightly patched CVS from a couple of months ago and the instructions that they provided (HOLD, dial extension, speak to said extension, then TRANSFER). Of course this isn't going to work nicely when you have more than one call on hold, so I'm looking forward to testing the patch that Frank Sautter wrote about on these lists some days ago! Bye, -- Emanuele ___ 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] ATXFER discussion, what's your opinion ?
[EMAIL PROTECTED] wrote: That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe because most asterisk PBX's are implemented using business class softphones rather than analogue phones? Most business class SIP phones (and even grandstream phones) allow attended/non-attended transfers with asterisk-stable and/or asterisk-head... I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi Oh, you mean the completely natural feeling put them on hold, dial new party, tell them you have a transfer, hit transfer? I want some of whatever kool-aid the person who thought that one up had. I still feel like I'm losing a call every time I do an attended transfer. Is there a _technical_ (e.g. SIP) limitation on this? ___ 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] atxfer features in stable release.
Hi all. At the end, i get atxfer with sip dowloading head cvs version of asterisk and this is ok, but now i have errors with h323. following the instructions i could compile h323 channel and load it, but when i call from sip to h323 or viceversa, i obtain this. debug - May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) -- H323/as5300-1.lpa.idec.net answered SIP/u0001-fbca May 4 12:12:07 WARNING[14186]: channel.c:2261 ast_channel_make_compatible: No path to translate from SIP/u0001-fbca(4) to H323/212.xxx.xxx.xxx(256) May 4 12:12:07 WARNING[14186]: app_dial.c:1315 dial_exec_full: Had to drop call because I couldn't make SIP/u0001-fbca compatible with H323/212.xxx.xxx.xxx == Spawn extension (default, 828111044, 1) exited non-zero on 'SIP/u0001-fbca' - end debug in the stable version, all its ok WHEN ATXFER AND THE REST OF FEATURESMAP FEATURES IN THE STABLE RELEASE? Best Regards¡¡¡ César García. Director de Sistemas, IdecNet S.A. Centro de Gestión de Red. Edificio IdecNet. C/Juan XXIII 44. E-35004, Las Palmas de Gran Canaria, Islas Canarias - España. Tfn: +34 828 111 000 Ext: 340 Henry Jensen escribió: Hello, I have 2 *, one is between a Siemens HiPath and the PSTN, having two PRIs connected to each side. When I call the Hipath to administer it (with Siemens HiPath Manager), I usually call through the PSTN and all wents well. However, I have a second Asterisk and when I call the first Asterisk trough the second to connect to the HiPath, the call comes not through. To show you what I mean: This works: HiPath Manager -- ISDN PBX -- PSTN -- Asterisk1 -- HiPath This doesn't work: HiPath Manager -- ISDN PBX -- Asterisk2 -- Asterisk1 -- HiPath Note: Voice calls are working perfectly, it's only the data calls that doesn't work. The debug output shows the following: -- Accepting unauthenticated call from XXX.XXX.XX.XX, requested format = 8, actual format = 8 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, Zap/g1/12345678) in new stack -- Called g1/12345678 -- Executing Dial(Zap/5-1, Zap/g2/12345678) in new stack -- Making new call for cr 32776 Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 8/0x8) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [...] -- Channel 0/1, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request I think the problem is the transfer capability: Speech line. It must be transfer capability: Unrestricted digital information. Is there a way to set the transfer capability? I noticed there is a file app_settransfercapability.c in CVS (but not in 1.0.7). Is this possible with IAX at all? Regards, Henry ___ 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] atxfer
Julian, Could you then tell me, from which version on atxfer is available? for instance does the 1.07 version from 19-3 support it ? I just don't like the idea of running bleeding edge code on my pbx :-) Kind regards, Joop Marijne On Fri, 25 Mar 2005 11:54:21 +0100, asterisk [EMAIL PROTECTED] wrote: I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) I am trying to get supervised/ attended tranfer working, blind transfer by pressing the # key works fine atxfer = * Attended transfers are only supported in CVS, not 1.0.X Julian J. M. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] atxfer
Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next week I am trying to get supervised/ attended tranfer working, blind transfer by pressing the # key works fine this is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 600 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 [featuremap] blindxfer = # atxfer = * I have tried atxfer=# and atxfer=*2 but nothing works, also, when I disable both blindxfer and atxfer, blind transfer still works. I have turned sip debug on, and I can see that asterisk does receieve the * keys parking calls does work btw. Kind regards, Joop ___ 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] atxfer
Look at the options for the dial command on the wiki, you have to use t or T or calls are not eligible to be transferred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Friday, March 25, 2005 3:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] atxfer Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next week I am trying to get supervised/ attended tranfer working, blind transfer by pressing the # key works fine this is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 600 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 [featuremap] blindxfer = # atxfer = * I have tried atxfer=# and atxfer=*2 but nothing works, also, when I disable both blindxfer and atxfer, blind transfer still works. I have turned sip debug on, and I can see that asterisk does receieve the * keys parking calls does work btw. Kind regards, Joop ___ 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] atxfer
On Fri, 25 Mar 2005 11:54:21 +0100, asterisk [EMAIL PROTECTED] wrote: I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) I am trying to get supervised/ attended tranfer working, blind transfer by pressing the # key works fine atxfer = * Attended transfers are only supported in CVS, not 1.0.X Julian J. M. ___ 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] Atxfer not working for called party
Hi. I've been trying to develop this module since some time now. CVS already has a dial version with atxfer. When trying this, using the modifiers tT and having configures features.conf accordingly, i havent been able to use such a feature in the called party. I also tried using t and T separately. I've tried to understand why this happens, and started to watch the copy of the frames in the bridge. I noticed only the frames originated from the calling party are analized for DTMF, dont know why When using stable asterisk, with normal Dial, the transfer function works ok with both parties. I Dont understand where is the problem, and if any have any ideas on how i can work it out, please tell me. Thanks in advance Bruce.- ___ 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
