similar to: Attended Transfer of a queue call fails

Hi Running bristuffed 0.3.0-PRE-1f which is 1.2.1. Using *2 in features.conf for attended transfer. Works well if someone answers. But the following sequence causes issue: 1. Receptionist takes call. 2. *2 then 123 to transfer to extension 123. 3. 123 is busy or does not answer so receptionist hears 123 voicemail 4. How can receptionist reconnect to calling user - could wait for voicemail to

Hi, recently I changend a few things in the configuration of the Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that different groups of SIP-Phones are using different trunks to the outside worls, so I moved some of them to a Support context. However, dial out from this phones failes as they're still looking for an extension in the default context, which doesn't

Situation such. There is an asterisk working as office pbx. 6 fxo - 18 fxs ports. All works perfectly, but some times in a week something occurs. Could not catch what exactly yet. But symptoms such. The asterisk infinitely writes the message of a type to broad gullies: WARNING [20757] chan_zap.c: We're Zap/8-1, not ... <ZOMBIE>. Numbers of channels can change. Because of that that broad

As I understood, Asterisk has a lot of features but lacks native 3-way calling and attended transfer. It would be great to have these features available to a simple IAX phone. I wonder how this could be implemented in Asterisk without asking for a patch. It should be possible with parking, conferencing, AGI and the manager interface. The extension 77 could be used by the attendant to blindly

Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP!

Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? -- Alexander Topolanek http://www.topolanek.at

Hi I have configured attended transfer in features.conf like this [general] parkext => 70 ; What ext. to dial to park parkpos => 00-99 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in parkingtime => 300 ; Number of seconds a call can be parked for

Hello, I'm running several asterisks in a carrier environment. The asterisks do mainly gateway business between E1 cards and IAX with some routing logic. On one key server I see issues of "Auto-congesting call due to slow response" coming every number of calls. The IAX peer is in the same subnet, the servers are not really loaded. Versions in use are 1.2.2 and 1.4.23-rc3, with rsa

Hi all, I'm new posting here, though not to perusing. I'm having an issue with attended transfer and was wondering if anyone had heard of the problem/had any suggestions... Apologies in advance if this post is excessively newb-oid. - An incoming call C is passed to A, a POTS telephone connected via a Handytone 286 ATA. - A presses atxfer key, then dials B, a Win XP laptop running

I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget

I upgraded my Asterisk system to version 1.2.14 to check if the sound quality issues I was having with Chanspy in 1.2.7 remained. I'm still getting them, and I'm honestly out of ideas except from RTFS. The called party sounds normally fine, but it's impossible to hear the caller. Sometimes, when the called party is talking, the caller can also be heard. The conversation sounds broken,

Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when

Hello, I have a problem with the Bristuffed version of Asterisk. I have tried Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have the same problem it seems: The setup: A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15. Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly. Grandstream gxp-2000 as a SIP phone, and a normal mobile

I need attended transfers, but I do not have time to talk to another extension and see if they accept the transfer, my features.conf is: [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 => 220 ; Number of

Hi, after upgrading to Asterisk 1.2.8 from 1.2.7.1 I got a problem with Grandstream BT100 after making an attended transfer (FLASH + NUMBER + SEND + WAIT ANSWER + TRANSFER). After the transfer, the display clears all the info except the clock, there is no dial tone, the WEB admin stops working. Incoming calls make the display light turn on but there is no ring and no callerid on the

Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus

I've searched for a few days now without finding an answer. The release notes for version 1.0.5.16 of the Grandstream firmware says it supports attended transfer using replace but the docs haven't been updated so I can't work out how to enable it, or whether it should Just Work. I'm currently using the # attended transfer patch for * but would like to get back to using the

I'm now up&running with - mISDN with avmfritz driver for Fritz PCI card - chan_capi from debian recompiled with a patch (see below) - EuroISDN with Point-to-Point (ptp) mode (Austria) - With Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k from debian sarge But am having some problems: 1) I needed to patch chan_capi.c from debian sarge (see below) to give the new channel to asterisk in state

I have a problem with Asterisk-bristuffed using a zaphfc card. I am located in the Netherlands, so I have an ISDN line from KPN. When I start Asterisk, and plug in the ISDN line, everything works perfectly for about 5 minutes. And then the ISDN line is down for 1 minute, and after that minute, the line comes back up and works for another 5 minutes. Every time the line goes down I get the error

Hi, I think I've identified an issue and just want to check before completing a bug report. Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code. If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works. Cases that do work are as follows... Calls using both Queue() and