similar to: Strange problem with latest Asterisk

Displaying 20 results from an estimated 100000 matches similar to: "Strange problem with latest Asterisk"

2005 Jan 05
1
Read() timeout hangs up the line
Hi list, I am having some difficulty implementing a certain dialplan where the following happens. If the first Dial() is not answered, I want to play a small greeting then ask the caller to either hold the line (try calling again) or press 1 to leave voicemail. exten => s,1,Dial(${BLAH},10,Tt) ; Dial 10 sec exten => s,2,Answer exten => s,3,Playback(greeting)
2008 Jan 09
0
[asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers
Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>
2003 Nov 16
1
strange Music on Hold between SNOM, Grandstream and Asterisk
Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 -> 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts
2006 Jun 14
0
Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!
I've got a strange situation with musiconhold. It works if I dial my extension 6000: >From extensions.conf: exten => 6000,1,Answer exten => 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer("SIP/gs1-b6ee", "") in new stack -- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack -- Started music on hold,
2004 Jun 25
0
3-way calling woes... Nasty static and inconsistent flash detection?
This is my setup: SPA-2000 -> Asterisk -> X101P (x4) -> PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on
2016 Nov 21
2
Asterisk 13.12.2 : strange queue behaviour
Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue at pbx-routing:15] Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on
2006 Jan 25
1
Want to automatically park call and have caller hear ring tones
Here's the short of it. I have an Asterisk 1.2.1 system setup to handle both personal and business calls. Now, the business callers will hear music while on hold, so the default MOH needs to play regular music. Personal callers should hear rings, not music. I have this working except for one specific case. If someone calls during the day (we're night people), asterisk will not ring
2004 Jul 23
0
qudBRI and transfering calls with the latest RC2.
I'm trying the latest bri 0.1.0 RC2 drivers. In announce I see implementation of so long waited Transfer feature. But I can't make it work. When the person who is making transfer after talking with second party press "R" second time to establish 3 way call the person to which call supposed to be transfered being disconnected. Any ideas whats wrong? Thanks, Dmitry
2010 Mar 01
0
Asterisk / Trixbox 2.6 Streaming MOH Problems
I've tried a number of solutions, but I've been unable to get Asterisk working with streaming MOH without running into the "buffer" issue. I've tried using various combinations madplay, mpg123, mpg321. I've also tried streamplayer by itself, and in combination with play-fifo ( http://www.freeswitch.org/asterisk_stuff/play-fifo.c ) to try and eliminate the issue. For
2006 Jan 25
0
Want to automatically park call and have callerhear ring tones
Short replay to long 'short of it'!! Use a queue for your calls set the queue to ring. 'r' option I belive. Set up a queue that has no members but allows you t 'joinempty' Setup an extension that AddQueueMember(home-silent). You will then need to hangup and the call will ring. Before entering the queue you could have the system send the YAC info for you. It will
2005 Mar 01
2
Park Craches asterisk
I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup
2006 Apr 29
0
canreinvite, bandwidth, dial option
I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a
2016 Nov 21
3
Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >
2004 Sep 10
0
chan_agent and SIP UA transfers fail
I am beating my head against a problem where queue calls offered by Agent channel to a SIP UA cannot be REFER transferred if the target UA/extension hasn't accepted the call. If the members of the queue are SIP channels, this is not a problem. I suspect chan_agent isn't flagging the bridge from Zap/n -> SIP/n properly, or this is by design. The following line is what is spoken before
2009 Feb 12
1
Problem with parking
Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state
2015 Mar 23
0
Question about hangup - Asterisk v11.15.0
Hello, on previous versions of asterisk, extension h and H make us know who ended a call (caller or callee). In the last * versions, seems that only h extension is used, as stated here http://www.voip-info.org/wiki/view/Asterisk+standard+extensions In the last versions, how do we know which end terminate a call (SIP, ISDN, Analog, ...) in h extension ? Will the
2004 Oct 05
0
Using Macro's that cause loops, on purpose and using h, exten in default twice
Please see my extensions below. I will try to type you through this. In default, an extension 5149053538 is matched on. This fires a macro that determines what time is it, and resets a variable which is then used to call another macro to place a call. if the call is answered, and the far end hangs up, the dial macro exits, then the routing macro exits, and you are back to default. at this point
2003 Sep 10
0
Transfer button on BudgeTone (Re: Transfer of queue call)
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 with a budgetone.. Later.. > With the right
2007 Jan 15
0
Parked calls with Asterisk 1.4.0
Hi List. We have a small issue with making parked calls work with the new Asterisk 1.4. I have an impression that this used to work with 1.2, so its either I'm doing something wrong, or a regression. I hope its not the latter and you can tell me what I'm doing wrong. The setup is an Asterisk with sip users in mysql realtime and dialplan in mysql static (mostly - some stuff is built-in).
2006 Mar 22
0
ZOMBIE on att transfer
I use asterisk 1.2.5 and h323 that comes with addons 1.2.1. Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and tries to make attendant transfer to person B (local SIP phone). They speak. Then A hangs up. Call form h323 trunk doesn't get to person B. This is what I get on CLI. -- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663 --