Re: [Asterisk-Users] Transfer from Grandstream BT100?

2003-11-03 Thread John Brown (CV)
what version of GS firmware are you running ? I call from PSTN to GS, GS does xfer to XTEN, hang up GS call continues if you aren't running 1.0.3.81 or newer, then upgrade :) john brown chagres On Mon, Nov 03, 2003 at 06:15:02PM -0600, Steven Sokol wrote: Hi, Does anybody know how to

RE: [Asterisk-Users] Transfer from Grandstream BT100?

2003-11-03 Thread Steven M. Sokol
I have 1.0.3.81. How do you execute the transfer? Thanks, Steven -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV) Sent: Monday, November 03, 2003 8:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Transfer from Grandstream BT100

Re: [Asterisk-Users] Transfer from IAX call

2003-10-07 Thread Dave Weis
On Fri, 3 Oct 2003, Richard Lyman wrote: you'll find that the context is being overwritten. look in chan_iax.c line 1628 and chan_iax2.c line 1645 (or within 3 lines of each) there is a sprintf that is stuff the context, if you comment those out, it should work again. Disclaimer: i have NO

Re: [Asterisk-Users] Transfer from IAX call

2003-10-03 Thread Richard Lyman
you'll find that the context is being overwritten. look in chan_iax.c line 1628 and chan_iax2.c line 1645 (or within 3 lines of each) there is a sprintf that is stuff the context, if you comment those out, it should work again. Disclaimer: i have NO CLUE what else this BREAKS!!! Dave Weis

[Asterisk-Users] Transfer fails periodically

2003-10-03 Thread jerk face
Has anybody else out there had a problem with transfers not being detected? Occasionally I will want to transfer somebody, so I'll hit the # key and instead of the Transfer application starting, the # tone is played. My hardware is T100P connected to an Adtran TA 750. I have relaxdtmf=yes in

Re: [Asterisk-Users] Transfer button on BudgeTone (Re: Transfer of queue call)

2003-09-11 Thread WipeOut .
The process for transfering a call with the Bugetone is as follows.. 1. Press transfer, you will get a dial tone.. 2. Dial in the extension to wish to transfer to.. 3. Press the Redial.. (on the newer phones this is the send button) You don't need the t option on your dial string to do transfers

Re: [Asterisk-Users] Transfer of queue call

2003-09-10 Thread Brian West
On the grandstreams if I recall the docs are incorrect on how the transfer feature works. Transfer + EXT + Transfer bkw On Tue, 9 Sep 2003, Hielke Christian Braun wrote: Hello, hope somebody can help. I have setup a queue which maps to some Budgetone SIP phones. When a call is answered,

[Asterisk-Users] Transfer button on BudgeTone (Re: Transfer of queue call)

2003-09-10 Thread Hielke Christian Braun
With the right extension setting, it works fine: exten = 1,1,queue,mainqueue|t I am not using the transfer button, but would love to. Right now i only have transfer with # key working. For example i have two BudgeTones registered to the *. Call from one to the other. Press the transfer

[Asterisk-Users] Transfer of queue call

2003-09-09 Thread Hielke Christian Braun
Hello, hope somebody can help. I have setup a queue which maps to some Budgetone SIP phones. When a call is answered, the # key to transfer a call does not work. Everything else regarding the queue works fine. Is there a way to activate it? Maybe something like the t option in the Dial

Re: [Asterisk-Users] Transfer of queue call

2003-09-09 Thread Richard Lyman
*CLI show application Queue -= Info about application 'Queue' =- [Synopsis]: Queue a call for a call queue [Description]: Queue(queuename[|options[|URL][|announceoverride]]): Queues an incoming call in a particular call queue as defined in queues.conf. This application returns -1 if the

[Asterisk-Users] Transfer (again!)

2003-09-05 Thread Daniel ANDRE
Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to

Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread WipeOut .
These are probably more issues for grandstream.. Maybe mail [EMAIL PROTECTED] with the issues about dropping both calls when the phone is hung up.. Later Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from

Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread William Zhang
GS phone does blind transfer only. Afer pressing transfer, you will hear dialtone and then dial the number, after dial the whole number, either wait more than 5 seconds or press redial/send button, then hangup, it should work. --- WipeOut . [EMAIL PROTECTED] wrote: These are probably more issues

Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread Daniel ANDRE
This works only if transfering to a phone wich is onhook. If it is off hook (busy), it doesn't work Is there any possibiliy to simulate transfert with dial plan? Regards, Daniel William Zhang a crit: GS phone does blind transfer only. Afer pressing transfer, you will hear dialtone and

Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread WipeOut .
The problem I thought he was refering to was that if phoneA is in a call with phoneB, then phoneA uses flash to put phoneB on hold and call phoneC then.. Problem 1 If phoneC hangs up then both the calls from phoneA to phoneC and phoneA to phoneB are disconnected. Problem 2 PhoneA has no way of

Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread Daniel ANDRE
It is exactly that and noway for PhoneA to connect PoneB and PhoneC each other. Daniel WipeOut . a crit: The problem I thought he was refering to was that if phoneA is in a call with phoneB, then phoneA uses "flash" to put phoneB on hold and call phoneC then.. Problem 1 If phoneC hangs

[Asterisk-Users] # Transfer context problem

2003-08-19 Thread John Fortman
Setup: Asterisk with chan_h323 (chan_iax was connecting the two clients directly, dropping asterisk out of the picture) Clients are two pentium class computers on the same network with ohphone installed. The idea is simply to have one client call into asterisk (a client calling from outside)

[Asterisk-Users] Transfer incomplete when MOH enabled

2003-05-30 Thread Marcus Adolfsson
Title: Message Potential Bug? CSV as of yesterday. Scenario: When Music on Hold is enabled, initiating a transfer from a Cisco 7960 using its built in transfer function (either transfer or blind transfer) to an analog phone on a TDM10B, the transfer is not sucessfull. The analog phone

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