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http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from Thirdlane) looks like a great program for "eye candy configuration" of Asterisk. However it costs lost of $, and I'm currently only an "experimenter" so to speak. Anyone advice of a decent alternative that is similar?? Currently, we only have VOIP connections, but will have a couple of Digium

...i add a custom extension in AMP with no dial string. Then add a dialstring in extensions_custom.conf like exten => 600,1,Dial(IAX2/username:password@host/${EXTEN}@from-internal) it works Bails -------------- next part -------------- An embedded message was scrubbed... From: bails <bails@westcomuk.com> Subject: using AMP custom extensions Date: Fri, 17 Feb 2006 16:25:16 +0000 Size: 1106 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20060217/35aa0d08/usingAMPcustomextensions.eml

...all, I've attached a copy of the debug from the Trend, if anyone cares to look. I'll probablt get more of a response fromt the list than the automated response from digium :( Thanks Bails -------------- next part -------------- An embedded message was scrubbed... From: bails <bails@westcomuk.com> Subject: Re: [Asterisk-Users] Sync Source: Internally clocked Date: Thu, 16 Mar 2006 10:59:15 +0000 Size: 4556 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/d1dfb6a5/Internallyclocked.eml

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), then it stop and come back all the way until the

I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to

So I have my extensions.conf (http://www.infiltrated.net/exten.stupidity.conf) shortened in case someone wants to look. Has someone encountered the following? I've racked my brain on this for too long... I have two contexts, day and night... Caller (Daytime) --> Dials an extension --> Caller hears extension ring on receiver --> Call goes through Caller (Night) --> Dials an

Hi I looking a good IAX service for a *emerging * voip provider. Better with a test account to try. Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte t?cnico ISPs Jabber ID: rpereyra@lugmen.org.ar For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 -------------- next part

Gents, Just a passing thought... is there any reason why the ability to search the past posts on here isn't switched on? Just wondered, since it makes much more sense to be able to search the old archives if you have a problem, rather than ask the same question again and again... Paul

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960

http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from Thirdlane) looks like a great program for "eye candy configuration" of Asterisk. However it costs lost of $, and I'm currently only an "experimenter" so to speak. Anyone advice of a decent alternative that is similar?? Currently, we only have VOIP connections, but will have a couple of Digium

Hi, Trying to set up these two buttons on a snom 360. The message waiting key seems to send a call to it's own number, which is obviously engaged and where you are prompted to leave another message to yourself, and the conference key seems to do nothing. Anyone manged to overcome these problems so that the conference key actually conferences, and the message waiting "retrieve"

Hi all I'm looking for a solution to this problem. *box--------internet-----------nat-----------softphone We have potential customers who will be travelling the world with laptops/pda's. They need to be able to connect to the asterisk box via ip wherever they are and will have no control over nat whatsoever. I have read that STUN offers this service, but cannot picture in my mind how

Are there any configuration options to allow certain sip/iax accounts to dial out over specific trunks, and also to stop them dialing out over other trunks. Thanks in advance Bails

Hi all, I have a small query regarding ringing tones on an iaxy2. I have a customer who uses an iaxy to breakout to pstn via our *. However the customer complains that he gets no ringing tone whislt making calls, i just visited the site and can confirm this. I also have another customer who is presently in canada with an iaxy calling thru our * , he doesnt have this issue. I presume that the

Hi I'm looking for a pinout for the above. Note this has what i'd call RJ45 sockets (or someone smart can correct me). I need to plug into BT (rj13?). And, yes I've googled (glad I'm not chinese) and have tried the suggested, just plug in a 6 connector rj11 and i didnt work atall. On a side note, I cannot beleive that Digium dont have this information on there site, it

We have 3 existing switches interconnected via dpnss, we need to integrate asterisk with these switches via a dpnss link. Any suggestions? also does anyone have a link to the differences between isdn30 and dpnss. Thanks in advance Bails

Hi all, I'm trying to setup a custom extension in AMP (yes i can code it by hand but the on-site admin that does moves & changes cannot). I've tried the following > add cutom extension 600 in the dial box i have Dial(IAX2/username:password@host/$EXTEN@from-internal) this doesnt work as these lines are added to extensions_additional.conf exten =>

Hi whatever I set the span line to in zaptel.conf ie span=1,0,0,ccs,hdb3,crc4 span=1,1,0,ccs,hdb3,crc4 span=1,2,0,ccs,hdb3,crc4 zttool always shows Sync Source: Internally clocked surely this cannot be correct.

Hi all I fancied playing with SER and * on the same box. So i thought i'd just change the default sip port for * in sip.conf [general] port = 5065 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) restarted * and now when i issue a > ]# netstat -anp |grep 5060 > udp 0 0 0.0.0.0:5060 0.0.0.0:*