similar to: WaitExten substitute

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

Hi everybody, I found in the company where I work an Antek EGW-804. I googled to see if it can be configured to work with * and I understood that it is possible, but I don't know how. Can someone help me? Thanks Stefano

Hi, In my dialplan I'm using a WaitExten() command. It works only with Zap phones. When I dial this command with Sip phone asterisk do nothing. Should I put extra definition in sip.conf to make this work with Sip phones? Thanks in advance Cheers

I am trying to use a WaitExten in a Macro, and I am finding that the extension which is pressed ends up in context of the calling context and not in the Macro. How do you do a WaitExten in a Macro? Tony Plack

Is WaitExten working in current CVS-HEAD? I'm attempting to use it in some new dialplan code but not getting anything.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/132b82c8/attachment.htm

Hi, all. My "autoattendant" looks like this: exten => s,1,Answer() exten => s,n,Background(corporate-greeting) exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds exten => s,n,WaitExten(30) When the call gets forwarded to the destination extension, however, there's no

Hello list, I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten. To simplify the problem I've brought it down to the following scenario: - 3 Asterisk Server A,B and C (central). - A and B both register with C. Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today!

I'm a bit confused about how WaitExten works. I assumed that when it returns 0, the next priority in the extension would be executed, but that doesn't seem to be the case. When I get to WaitExten and enter extension 8, it plays the message, then Waits another 10 seconds and times out. [local] exten => s,1,Wait,1 ; Wait a second, just for fun exten =>

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

Hi list, I am ready to implement VoIP for 2 of our offices. i do like to setup asterisk box in each office (behind PBX) and use our ADSL links to provide the tunnel. This will provide cheap telephone calls between offices. I also like to setting PBX, so it can let incoming call-thourgh to asterisk extensions, so our home users can use office * box to call to other office. I don't want to

I have just tarred up a snapshot and uploaded it to: http://www.mindrot.org/misc/openssh/openssh-SNAP-20000823.tar.gz The snapshot incorporates the last month's fixes and enhancements from the openssh-unix-dev mailing list and from the OpenBSD developers. In particular: - ssh-agent and ssh-add now handle DSA keys. NB. this does not interop with ssh.com's ssh-agent. (Markus Friedl)

>From samles config. When user change uavarible message from firefly login by IAX got 3 file .WAV .gsm . wav But no sound record. When login by antek 804 GW (SIP mode) record success. but message can't play when someone need to leave voice mail to this box. But when delete .WAV . gsm and leave only .wav its' work. Is problem in asterisk Or client Dome C.

Sorry my bad, thanks for spotting it. Should that explains also the failure to grab the mutex? Andrea Il 3/12/2019 12:14 PM, Rowland Penny via samba ha scritto: > On Tue, 12 Mar 2019 12:01:08 +0100 > Andrea Cucciarre' <acucciarre at cloudian.com> wrote: > >> The OS is OmniOS, the DC is Windows Server (not sure about the >> release), and below the smb.conf.

When tar'ing up an ext3 filesystem on a 3ware RAID I have, I got the following errors in my system log: Apr 7 21:27:46 allo 3w-xxxx[957]: Drive error encountered on port 3 on controller ID:0. Check cables and drives for media errors. (0xa) Apr 7 21:27:46 allo kernel: 3w-xxxx: scsi0: Command failed: status = 0xc7, flags = 0x51, unit #0. Apr 7 21:27:58 allo last message repeated 3 times Apr