similar to: Dialplan end of pattern matching question

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

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

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, 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?

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

I want to build an application that has the concept of administrative domains. What I mean by this is that administrators have access to different data, based on what domains they are a member of. The domain strucutre is hierarchical. Here is an example: - MLB - AL - East - Yankees - Red Sox ... + Central + West - NL + East + Central + West Now

I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct

Hi everybody. I want to identify duplicate numbers and to increase a value of 0.01 for each time that it is duplicated. Example: x=c(1,2,3,5,6,2,8,9,2,2) I want to do this: 1 2 + 0.01 3 5 6 2 + 0.02 8 9 2 + 0.03 2 + 0.04 I am trying to get something like this: 1 2.01 3 5 6 2.02 8 9 2.03 2.04 Actually I just know the way to identify the duplicated numbers rbind(x, duplicated(x) |

Hello there. Happy new year for everyone! I need help with a table. This table contains 300 rows and 192 columns. Being the first column the ID of my samples that can have several observations. I need to generate e NEW table that contains a single ID with the sum of the observations by columns: For example: Example ID?? A??? B? ? C? ? D? ? E? ? F? ? G.... 191 columns a1?? 0??? 0??? 0? ? 1???

Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the

i am trying to setup a speed dial application in my dialplan, but whenever i try to dial the pattern matched extension i get a 404 not found. my extension is defined as '_XX#,1,goto(default,${DB(${CALLERIDNUM}/${EXTEN:0:2})},1)'. after doing some testing, i added an extension defined as '_.,1,NoOp(!* dial ${EXTEN})' and removed everything else in my context. i found that the

I'm using Polycom Soundpoint phones and I want to use some extensions beginning with # for features setup. I'm getting the fast busy "can't match it" signal. I want to match #50 for call forwarding, for instance, and #505551212 to set the call forwarding number and turn it on. I have tftp set up and sip.cfg contains the following: <dialplan

Hi Is there anyway to add logic to dialplan pattern matching? I would like to match all toll free numbers with one pattern, so 1800, 1877, 1866, 1855, etc. I can't figure out how to do this in dialplan syntax. As a programmer, I want to say 18[00 or 77 or 66 or 55 etc]. Can't figure out if this is even possible with dialplan pattern matching (though I suspect it is somehow). Andrew