similar to: How to use return value in extensions.conf

Hi all, Maybe I'm being thick here, but I've had a look through the mailing list and the Wiki, and I can't seem to see details of anybody else with this problem.... With the following line: exten => s,1,HasNewVoicemail(201) I am getting the following error: -- Executing HasNewVoicemail("SIP/201-2f1e", "201") in new stack Sep 2 12:41:09

Hi, it seems if a user leaves voicemail and hangs up the call when done, then HasNewVoicemail never gets called on the next line in the context. However if they press # to finish their VM, then it moves to HasNewVoicemail and this works? eg:- ...... exten => 2002,3,VoiceMail(u${OFFICEVM}) exten => 2002,4,HasNewVoiceMail(2002) ...... exten => 2002,105, do something cos vm has been

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Hi, for some weeks now I have been unable to make calls via my voicepulse connect IAX account? When I attempt the console looks like this:- rt*CLI> -- Executing Dial("SIP/2008-cf55", "IAX2/NBhXXXXXX:XXXXXXN82@gwiaxt01.voicepulse.com/12124565900") in new stack -- Called NBhXXXXX:XXXXN82@gwiaxt01.voicepulse.com/12124565900 -- Call accepted by 66.234.228.160

Hi all, I have a demo machine I'm running up on Lenny - it has the packaged Asterisk version installed (1.4.21.2+stuff). I'm trying to add an extension to leave a voicemail message, just with Voicemail(1234), which I've done before (on 1.2 at least), but it's saying "no application 'Voicemail' ". "module show like voi" shows

Hi I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. 302 => 302,Website Sales,sip@example.com,,attach=yes|delete=yes However I can't seem to find a way to test is a message is left. I have tried

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

Thanks, Anthony. I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup code is run by both the caller channel and the peer channel, but I only want the caller channel to do that. Also, when the peer hangs-up, there is no execution of the priorities following the Dial. Finally, is there a way to reset all globals, maybe as a variant of "dialplan

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 733 +++++++++++++++++++++--------------------- 1 files changed, 370 insertions(+), 363 deletions(-) diff --git

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 733 +++++++++++++++++++++--------------------- 1 files changed, 370 insertions(+), 363 deletions(-) diff --git

Rename camel case variables in channel.c and changed them to lowercase. Sending this from my own accounts till we have a proper mail server set up that allows us to send out patches without issues. This patch was created by Haiyang Zhang. Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> ---

Rename camel case variables in channel.c and changed them to lowercase. Sending this from my own accounts till we have a proper mail server set up that allows us to send out patches without issues. This patch was created by Haiyang Zhang. Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> ---

i am developing a web application to manage callcenter, i will shortly release it on sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i use this commands with manager API: Action: Redirect\r\n Channel: SIP/200-sads\r\n ExtraChannel:

Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the "Idle" state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him

OK here is a possible tricky one. I have a rocom door entry system which connects to an FXS port on my TDM400P card. When the door button is pressed it initiates an 's' extension which dials a number of SIP extensions. When a SIP phone is picked up the user can speak to the person at the door and press the 7 digit which sends at DTMF tone to the rocom unit opening the door. All this

Hello All, Here is a patch that fixes the problem when forwarding messages with vmail.cgi. Bug submitted with patch on bugs.digium.com. -sb --- /usr/src/asterisk/vmail.cgi.orig 2003-12-17 14:21:47.000000000 -0500 +++ /usr/src/asterisk/vmail.cgi 2003-12-17 15:07:36.000000000 -0500 @@ -672,7 +672,7 @@ sub message_copy() { - my ($mbox, $oldfolder, $old, $newmbox, $new) = @_; + my ($mbox,

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 729 +++++++++++++++++++++--------------------- 1 files changed, 368 insertions(+), 361 deletions(-) diff --git

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 729 +++++++++++++++++++++--------------------- 1 files changed, 368 insertions(+), 361 deletions(-) diff --git