similar to: Call Queuing

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same

Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a "900 number". Effectively the caller pays much more to call such a number than a normal national or local call. The problem with these is that I don't want Asterisk to actually signal to the

The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 14.4.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect). When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

The Asterisk Development Team would like to announce the release of Asterisk 17.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

Hi- Perhaps someone with an E100P in hand can answer this: I just received an E100P from Digium (I normally buy quad boards) I noticed that the circuit board says "T100P" on it, and I assume that the T100P and E100P both use the same circuit board. Can someone please confirm that their E100P says "T100P" on the artwork? Thanks Scott Scott M. Stingel Emerging Voice

I'm working with * and queuing and while things are mostly-working, they don't work quite as well as the docs on the wiki indicate they should. Things like leavewhenempty, the h option to queue, stuff like that. I search the archives and it seems that very few of the queuing questions are answered on the list. I'm curious why that is. I figure there are X possibilities: 1. The info

Hello, Thanks to members of the list things changed on my installation (asterisk + digium E100P with of course an E1 line). Here is my zaptel.conf : - span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr - Output of zttool: - Alarms Span YEL/RED Digium Wildcard E100P E1/PRA Card 0 - [root@asterisk root]# head /proc/zaptel/1 Span 1: WCT1/0 "Digium

Just cross pairs 1-2 and 4-5 1 <--> 4 2 <--> 5 Never done an E1, but I think it's the same for a T1 4 wire cable. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Monday, December 29, 2003 9:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] E100P pinouts

Hello, I have a E100P card from digium and I try to implement a conference bridge in asterisk. I wonder since I got the E100P card do I still need to load ztdummy for caller from h323 endpoints to work with Meetme? I load the E100P driver but i did not load the ztdummy driver. My h323 caller does not hear any voice play by Meetme. Looks like ztdummy is required as long as h323 is concern and

i have a question with queuing discipline " if i create many classs with sfq and cbq and ... i would like to know about qos . which queuing discipline will be chosen first. ( not use priority ) (queuing discipline) sfq---------> packet incoming packet outgoing cbq--------> Thank you .

The Asterisk Development Team would like to announce the release of Asterisk 13.15.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 14.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.23.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi, I had a question regarding asterisk installation with 2 E100P card. I would like to do such installation but I had some troubles: PABX -> E100P -> asterisk -> E100P -> PSTN In fact I would like to add some IP Phones to my PABX wich is full. But with this installation, I can't place call. I can receive call without problems. I also can place call from IP Phone, but I