Re: [asterisk-users] Hook Flash
Most ATAs have the capability of sending hook flashes to the server. That's how all mine are configured, and Asterisk handles hook flashes. As of 18.4/16.17, there is a Flash AMI event as well that can be used to listen for these and do something configurable. On 6/25/2021 2:41 PM, Telium Technical Support wrote: > > Since this function is handled by the ATA, you would have to look > there (or post details) for something ATA specific. In general I > don’t think so, hook flash just puts one channel on hold a > creates/answers another. But, you may be able to script the > functionality you need it in the Ast dialplan. > > > > *From:*asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] > *On Behalf Of *Dovid Bender > *Sent:* Friday, June 25, 2021 3:26 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* [asterisk-users] Hook Flash > > > > Hi, > > > > It's been a very long time since I dealt with a along lines. Does > anyone know if there is a way to "pass though" a hook flash? I am > working on a project where there will be one FXS and one FXO. I want > if there is call waiting for the phone connected to the FXS to be able > to hit the hook and have that sent back out on the FXO port. > > > > TIA > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hook Flash
Since this function is handled by the ATA, you would have to look there (or post details) for something ATA specific. In general I don’t think so, hook flash just puts one channel on hold a creates/answers another. But, you may be able to script the functionality you need it in the Ast dialplan. From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Friday, June 25, 2021 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Hook Flash Hi, It's been a very long time since I dealt with a along lines. Does anyone know if there is a way to "pass though" a hook flash? I am working on a project where there will be one FXS and one FXO. I want if there is call waiting for the phone connected to the FXS to be able to hit the hook and have that sent back out on the FXO port. TIA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hook Flash
Hi, It's been a very long time since I dealt with a along lines. Does anyone know if there is a way to "pass though" a hook flash? I am working on a project where there will be one FXS and one FXO. I want if there is call waiting for the phone connected to the FXS to be able to hit the hook and have that sent back out on the FXO port. TIA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hook Flash
Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel 1.4.11 to a Panasonic PBX. I'm using dynamic features to send hook flash to the zap channels to make a call transfer to the pbx without tying a channel. When I call from asterisk to the Panasonic PBX I haven't any no problem, but when the call is from the Panasonic PBX, the dynamic features doesn't work. I have already tried all possible combinations in feature.conf: zapflash = *3,peer/both,flash zapflash2 = *4,callee,flash zapflash2 = *5,caller,flash In all cases I am setting the variable DYNAMIC_FEATURES before the Dial(). And is not a dtmf problem because I can see in the console the debug of the DTMF: chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '*' chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '*' chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '3' chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '3' The problem is that the application mapped in feature.conf it isn't been triggered. I would appreciate any help, I have already googled to death and I couldn't find anything. Thanks in advance. Lucas Alvarez -- ___ -- 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] Hook Flash
- Lucas Alvarez [EMAIL PROTECTED] wrote: Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel 1.4.11 to a Panasonic PBX. I'm using dynamic features to send hook flash to the zap channels to make a call transfer to the pbx without tying a channel. When I call from asterisk to the Panasonic PBX I haven't any no problem, but when the call is from the Panasonic PBX, the dynamic features doesn't work. I have already tried all possible combinations in feature.conf: zapflash = *3,peer/both,flash zapflash2 = *4,callee,flash zapflash2 = *5,caller,flash In all cases I am setting the variable DYNAMIC_FEATURES before the Dial(). And is not a dtmf problem because I can see in the console the debug of the DTMF: chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '*' chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '*' chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '3' chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '3' The problem is that the application mapped in feature.conf it isn't been triggered. I would appreciate any help, I have already googled to death and I couldn't find anything. Thanks in advance. Lucas Alvarez -- Perhaps it is a matter of how fast the DTMF is being delivered from the other PBX. You can adjust the featuredigittimeout in features.conf to see if that is the case. Jeff ___ -- 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] Hook flash time problem on TDM400/FXS
I have been trying for some time now to make the hook flash work on the FXS port. I am using Asterisk 1.4.10.1 with zaptel 1.4.4. When I manually flash the hook I can manage to find the duration to put a call on hold. However when pushing the flash button it never works. The phone's flashtime seems to be too short. I tried to set a shorter flashtime in the zapata.conf file, but it seems to be ignored. I have flash=100 configured in the zapata.conf en when reloading it, this is what is reported on the asterisk console: == Parsing '/etc/asterisk/zapata.conf': == Parsing '/etc/asterisk/mgcp.conf': Found Found [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring flash [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 1, FXS Kewlstart signalling [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 2, FXO Kewlstart signalling [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 3, FXO Kewlstart signalling [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 4, FXO Kewlstart signalling How can I adjust the flashtime on a FXS port in such a way that I can use the flash function of the phone connected to it. This is especially important for a DECT phone for which I cannot do a manual hook flash. Without this I cannot transfer calls. My searches on the internet did not give me any information other than that I should change the flash time parameter in zapata.conf. An old message indicated that I should change some .h file and recompile zaptel drivers, but could not find the particular piece of code probably because it has changed considerably later on. Also the information I found indicated that the flash time in Europe is generally very short (80 - 120 ms) as compared to the US (750ms ?). I am in the Netherlands. TIA, Hans Feringa ___ --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] hook flash call transfer
I am trying to use hook flash to transfer a call but I want the recording on the line I transfer to to start after I hang up. In other words if I receive a call and want to transfer it to VM or to a recording, I want to be able to flash the hook, dial the extension, and hang up. But I do not want the recording/vm message to satrt until the call is actually transfered. Is this possible? My work around it to insert a wait in the beginning of the contect I am transferring to. Is there a cleaner way? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.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] Hook Flash via SIP INFO command?
Is there a way to signal a hook flash via sip info to have a Sipura 3000 or other non-zaptel FXO flash hook for CW / PBX integration? Bryan McLellan [EMAIL PROTECTED] Strategy Computers, Inc. 2475 140th Ave NE, C-100 Bellevue, WA 98005 425-643-4849 Fax 425-643-4854 ___ --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] Hook-Flash on Voicetronix
I'm using a Voicetronix OpenSwitch12 with Asterisk. I need the ability to hook-flash a channel while a caller is on the line so I can transfer a call through a PBX.I need the equivalent of theFLASH() application, which only works with Zap channels. Does anyone know how I can hook-flash then send DTMF on a vpb channel? Rodney Hassell ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
For threeway calling (analog phone) I just hit the flash button get a dial tone, dial the number and hit the flash key again. It doesn't work for me when I'm using asterisk. No problems without it. So is my hardware broken or my dialplan? When you hit the flash key is anything displayed in the CLI ? A while back, someone posted a list of built-in extension numbers that are built into the zap channel module. The list included: *0 Send hook flash *67 Disable Caller ID *69 Say last caller's Caller ID *70 Disable call waiting *72 Activate call forward immediate *73 Deactivate *78 Enable Do Not Disturb *79 Disable Do Not Disturb *80 Add last caller's caller ID to blacklist *82 Enable Caller ID (only if disabled with *67) I don't use the above, but they certainly appear to be the ones your looking for. Obviously some of the features noted in that list do not exist in asterisk, therefore it would suggest they apply to the pstn/zap interface . That same posting indicated the above extensions could be overrode with other entries in extensions.conf. ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. I've tried using combinations of hook flash button and *0 on three different phones and I dont get a dial tone, the other party is not put on hold, and I don't see the keys I'm pressing in the CLI. When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. Can someone post an example of a working extensions.conf / zapata.conf where they use hook/flash that I can try. ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
On December 29, 2004 06:25 pm, PHP Mechanic wrote: Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. I've tried using combinations of hook flash button and *0 on three different phones and I dont get a dial tone, the other party is not put on hold, and I don't see the keys I'm pressing in the CLI. Are you trying to use these features in * or on the line? When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. Can someone post an example of a working extensions.conf / zapata.conf where they use hook/flash that I can try. This sounds like you are subscribed to these services via your telco -- this means you need to flash the line, not your phone. To do something like that I imagine you'd have to hit # or hookflash your phone and then have dialplan logic in extensions.conf which would Flash() the proper Zap line. Doesn't sound easy but I've never done it myself. -A. ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
For threeway calling (analog phone) I just hit the flash button get a dial tone, dial the number and hit the flash key again. --- PHP Mechanic [EMAIL PROTECTED] wrote: Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. Are you trying to use these features in * or on the line? I'm trying this on my analogue phone that are connected to asterisk via the tdm411b. I can see I have call waiting and can't do anything about it - pretty frustrating. When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. To do something like that I imagine you'd have to hit # or hookflash your phone and then have dialplan logic in extensions.conf which would Flash() the proper Zap line. Yes, I figure I probably have to do something like this. Can someone post an example of how they do it. or show me a dialplan of how I can transfer a connected caller to a conference room to achieve the same thing etc. ___ 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 __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hook/Flash, Hold, Call Waiting, Three Way Calling
For threeway calling (analog phone) I just hit the flash button get a dial tone, dial the number and hit the flash key again. It doesn't work for me when I'm using asterisk. No problems without it. So is my hardware broken or my dialplan? When you hit the flash key is anything displayed in the CLI ? ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
On December 29, 2004 07:05 pm, Richard Reina wrote: For threeway calling (analog phone) I just hit the flash button get a dial tone, dial the number and hit the flash key again. You're missing the point. POTS - Asterisk - Analog phone He's got call waiting/threeway calling on his POTS line -- Asterisk has no way of passing this on to the phone outside of the audible beep you hear. The best thing I can think of for him is something like this *1,1,Flash(Zap/1) So when he hears the beep, he hookflashes, hits *1 and is rejoined... I have no idea if it'd actually work or not though, since I have no phone line at home. :-) -A. ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
PHP Mechanic wrote: Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. I've tried using combinations of hook flash button and *0 on three different phones and I dont get a dial tone, the other party is not put on hold, and I don't see the keys I'm pressing in the CLI. When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. Can someone post an example of a working extensions.conf / zapata.conf where they use hook/flash that I can try. Do you have the following in your /etc/asterisk/zapata.conf BEFORE the channel number? ; ; Support three-way calling ; threewaycalling=yes ; ; Support flash-hook call transfer (requires three way calling) ; transfer=yes ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. I've tried using combinations of hook flash button and *0 on three different phones and I dont get a dial tone, the other party is not put on hold, and I don't see the keys I'm pressing in the CLI. When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. Can someone post an example of a working extensions.conf / zapata.conf where they use hook/flash that I can try. Do you have the following in your /etc/asterisk/zapata.conf BEFORE the channel number? Yes. My zapata.conf is below. My setup is POTS - Asterisk/TDM411B - PSTN Line. While connected to another party if I press flash or *0 on my analoge phone I don't get a dialtone, the called party does not go on hold and I don't see anything in the CLI - it doesn't work. It all works fine when I don't use asterisk. [channels] ; ; TDM400P Port #4 plugged wall ; This is the PSTN Line ; context=PSTN signalling=fxs_ls busydetect=yes ; to test when a line is hung-up busycount=6 ; to prevent suprious hangups echotraining=800 echocancel=yes immediate=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ; ; TDM400P Port #1 plugged into analog Phone ; This phone is allowed to dial extensions and local and long distance numbers ; context=RealPhone signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Real Phone 1 mailbox=1 channel = 1 ___ 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] Hook-flash timing
Finally, can I turn off the '#' to transfer, since we're using the hook-flash (albeit manually) instead? ISTR an option to do this but have spent the morning trying to find it again unsucessfully... I think you might want to look at the 'T' and 't' options on the Dial application, documented somewhat here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial jl ___ 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] Hook-flash timing
On Wed, 2004-08-04 at 01:46, john lawler wrote: Finally, can I turn off the '#' to transfer, since we're using the hook-flash (albeit manually) instead? ISTR an option to do this but have spent the morning trying to find it again unsucessfully... I think you might want to look at the 'T' and 't' options on the Dial application, documented somewhat here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial This subject relates to a problem I am having, though likely it is different. I am using digium TDB40B connected to handsets, and I can't use the flash/recall button on the phones. However, I can 'manually' do the hook-flash. From memory (not good) I recall that in Australia the phones use a shorter flash time, so I added the rxflash to my zapata.conf file. Using different values, I managed to get one of two possibilities: 1) Press flash, and the other party hears the DTMF tone. No Transfer. 2) Press flash, the other party is dis-connected (hears busy/congestion tone) and you get a new dial-tone. As if you had hung up for a few seconds, and then picked up to make a new call. (1) happened if rxflash was higher than about 103, while (2) happened with any number less than that. I recall there being a number of parameters that need changing to make this work, including reducing the pulse dial time, etc. However, I haven't been able to find this in google/wiki. Does someone perhaps know the magical answer to this? Thanks, Adam ___ 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] Hook-flash timing
Hi, Is there any documentation on the fields prewink, preflash, wink, flash, rxwink, rxflash, start and debounce in zapata.conf? The Recall button on my phone doesn't seem to trigger a transfer via my shiny new TDM40B. However, tapping the hook does, but only if I tap it for long enough. Presumably the Recall button's timing is too short? Further, most users who press on the hook to hang up and start a new call are not holding the hook down for long enough and are in fact starting a transfer instead. Presumably the fields above are what I need to change to get all the timings shorter? I've spent a while trying and have managed to have an effect but I still can't work out what the fields mean. Pressing Recall at the dialtone makes Asterisk think I dialled 1 (presumably using pulse). Do I just need to disable pulse dialling detection? How do I do this? Finally, can I turn off the '#' to transfer, since we're using the hook-flash (albeit manually) instead? ISTR an option to do this but have spent the morning trying to find it again unsucessfully... Cheers, Robie. ___ 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] Hook Flash INFO messages
Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with aSIP Proxy/Softswitchother than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature asVoIP gateways, saw the hookflash and passed a SIP INFO (event hookflash) back to the Proxy. The Proxy sent this message on to the calling SIP phone which replied that this "feature is not implemented." The gateway manufacturer says that theProxy should process the INFO packet, place the calling endpoint on hold (as a PBX would), stream dialtone to the gateway prompting the user to dial the digits indicating the destination to whom the calling party should be transferred, and then do a transfer. The Proxy manufacturer says that the gateway should see the hookflash,Hold the caller locally (as a SIP phone would), and give new dialtone to the single line phone prompting the user to dial the digits digits indicating the destination to whom the calling party should be transferred, and then send a complete transfer sequence to the Proxy. My question is, how would Asterisk handlea situation like this? Are there any opinions as to how this scenario should be handled? Sean
Re: [Asterisk-Users] Hook Flash INFO messages
On Fri, 2003-07-11 at 22:12, Sean P. Robertson wrote: Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways, saw the hookflash and passed a SIP INFO (event hookflash) back to the Proxy. The Proxy sent this message on to the calling SIP phone which replied that this feature is not implemented. The gateway manufacturer says that the Proxy should process the INFO packet, place the calling endpoint on hold (as a PBX would), stream dialtone to the gateway prompting the user to dial the digits indicating the destination to whom the calling party should be transferred, and then do a transfer. The Proxy manufacturer says that the gateway should see the hookflash, Hold the caller locally (as a SIP phone would), and give new dialtone to the single line phone prompting the user to dial the digits digits indicating the destination to whom the calling party should be transferred, and then send a complete transfer sequence to the Proxy. My question is, how would Asterisk handle a situation like this? Are there any opinions as to how this scenario should be handled? Asterisk currently only handles dtmf INFO messages. --Karl Sean -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
