similar to: channel var for Call on hold?

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the

Hi, Does anyone had the problem of asterisk SendFax + spandsp sending only the first page of a multi-page TIFF file? Seams to be related to spandsp ECM config. Any thoughts about it? Thanks, Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121005/ac471600/attachment.htm>

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

Hi all, Does anyone uses astDB for a large amount of data, in special for implementing black lists with millions of numbers (i'd like about 2 or 3 million)? That would be held in memory right? Is this (memory consumption) the only problem I could face? Att. Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've noticed an unusual behavior on the dialplan execution: assume this DP: exten => _6XXX,1,NoOp(test1) exten => _XXXX,1,NoOp(test2) exten => _XXXX,2,NoOp(test3) If I call 6000 then test1 and test3 NoOps get executed, even though the pattern is different. I've always thought that if I call 6000 it would match the 6XXX pattern, that only has 1 priority, that would get

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

Hi all, After * starts the command "queue show" would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any "QueueMemberStatus" for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a

Hi all, When I have 2 masks that would like to execute the same logic, there is the way to use the Goto (or any other) command without changing the ${EXTEN}? Eg. DID range is 1200-1349 -> call Macro(disca), what mask to use? (I just got it with 2 masks, but I didn't wanted to duplicate the dialplan for both) [test] exten => _12XX,1,Set(DIR=3) exten =>

Hi all, I'm trying to use the CDR() function on the "Variable" header of the Originate AMI action, but it isn't working. Anyone knows anything about this problem? asterisk 1.4.26 Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

Hi all, I'm trying to send an email with the voicemail details and I want to send a HTML link on it to make a click2call to the voicemail main, but the email is send with 'text/plain' encoding and thus it will not show the link, but the HTML in plain text on the body of the email, How can I change the enconding to 'text/html' so the link will get displayed correctly?

Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten <-> exten calls, and not for outbound calls -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090117/a53f3178/attachment.htm

This sure seems like it would be simple. Probably can't see the forest for the trees. I need to use the "native" MOH feature on my little WRT to save processor load. I normally don't use MOH but am playing with atxfer and would like to have something to play to the remote transferee. But when I comment out the "default" clause in musiconhold.conf, I get an error

Not holding the mutex potentially causes corruption of the kernel channel when page flipping. Cc: stable at vger.kernel.org #3.13 Signed-off-by: Maarten Lankhorst <maarten.lankhorst at canonical.com> --- diff --git a/drivers/gpu/drm/nouveau/nouveau_display.c b/drivers/gpu/drm/nouveau/nouveau_display.c index 29c3efdfc7dd..76e3cf025c10 100644 --- a/drivers/gpu/drm/nouveau/nouveau_display.c

Hello, I am using asterisk manager interface (http) for originating calls. How can I get the name of the channel which is created by originate? I want to use this channel for other manager commands like Atxfer, Monitor, Hangup etc. If I do action=originate, channel=SIP/200 then it creates a channel like 'SIP/200-0865ff80' which I can see in the asterisk console using "core show

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font face="Tahoma">Hi Everybody,<br> <br> I&acute;m getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ... In some calls, i get an atxfer or musiconhold in the middle of

Hi, I've created a tiny dialplan to test the return of a call on transfers, like this: (I had to use the DEVSTATE backport here) [phones] exten => _12XX,1,Dial(SIP/${EXTEN},6,tT) exten => _12XX,n,GotoIf($[ "x${BLINDTRANSFER}" = "x" ]?noBT) exten => _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)}); exten => _12XX,n,Goto(dRet) exten => _12XX,n(noBT),GotoIf($[

When I restart Samba, and then connect to it from an XP screen, I get the following message in the /var/log/messages file: Dec 11 16:36:38 yavin smb: smbd shutdown succeeded Dec 11 16:36:39 yavin smb: nmbd shutdown succeeded Dec 11 16:36:39 yavin smb: smbd startup succeeded Dec 11 16:36:39 yavin smb: nmbd startup succeeded Dec 11 16:36:48 yavin samba(pam_unix)[1742]: session opened for user

Hi Gabriel, Yes this information is shown in real-time and also in historical reports with the OrderlyStats system. OrderlyStats is now available as a Server Edition you can download and install yourself, as well as the FREE managed service. You can get it at http://www.orderlyq.com/statistics.html Hope this helps, Matt. Gabriel Ortiz wrote: Hi all, Is there a way to get the time

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900