similar to: SIP Dialplan Failover Solution

Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don't require it) can cause problems? Regards and thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there ..... ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, I think I've discovered a little bug in t.38 bug in 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. Asterisk always responds with a=T38MaxBitRate:2400. I've tried with Patton and Grandstream devices and the result is always the same. Patton ignores the parameter and sends the fax at 9600.

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone number for few seconds, checks if it "hears" the ringing and hangs up the call. ** I use

Hello, For lab testing, I'm trying to build two differents PJSIP trunks between two Asterisk 13.8.0enabled boxes. I thought I could set up both trunks like this: Box A/port 5060 <------ Trunk1 -----> Box B/port 5060 Box A/port 5062 <------ Trunk2 -----> Box B/port 5062 and declare trunks like this: [foobar1] type=endpoint transport=simpletrans context=from-customer

Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there. I've done some googling around to try and find an example of what I'm trying to do, but it's one of those things that just seems hard to find the right terms to search for. If there's some documentation out there on this, I'd appreciate being pointed in the right direction. If not, then if someone has some

Hello, I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to providers Cisco 2800 with VWIC-1MFT-E1 card. the same card runs fine with another E1 provider. TE420 led's lite green. Message type: RELEASE COMPLETE (90) < [08 02 80 ac] < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) < Ext: 1 Cause:

Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding

Hello guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005 at 80.75.132.66 trunk2: 73432260050 at 80.75.132.66 Thing is I can?t figure out how to route them to different IVRs by default Asterisk can?t match endpoint Request from '<sip:+ 73432260005 at 80.75.132.66>' failed for '80.75.132.66:5060' (callid:

Hi, I have IVR and when I press 1, asterisk calls my mobile phone. If my mobile phone is offline, asterisk transfers to asterisk voicemail. I'd like asterisk detects my mobile voicemail and if my mobile voicemail answers, asterisk transfers to asterisk voicemail. For that, I used AMD. So I have problems ! Asterisk detects answering machine everytime! How do I do please ? extensions.conf

I just thought i'd post a small sample that uses the new chan_local to show one way of doing variable callfwding This sample extension.conf uses's the ast DB to store a users current extension, in a db family of CallFWD and the unique Key is based on the current channel the user is assigned. In the globals var section each key is hardcoded EXT1, EXT2 this is used in the [incoming] context

Is there any way to have users be able to turn on or off call forwarding at the asterisk server, so they can configure their own forwarding number and enable/disable it? Hopefully, with the added benefit that it will remain on between server reloads and restarts? I have written a hack -- a AGI script to do various checking, and if the destination is "ok" set a database variable

Hello, I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw] include = from-didww [from-didww] exten = 3130XXXXXXX,1,AGI("did.php") exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20) but when i run the rule it says chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension

Before I start, I just want to say this is not necessarily a problem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 Tollfree Trunk1 0 0 100 1416 Canada Trunk2 0 0

Hi, I want to do so that extension 501 will always use trunk1 for outbound calls and 502 will use trunk2 for outboud calls. How do I do this. Right now all extensions use the same trunk for outbound calls. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there anyway to get the following scenario to work... I have 3 IAX trunks that I want to setup to peer with other * boxes. I have 1 physical interface, eth0. I also have 2 sub interfaces, eth0:1 & eth0:2. I want to setup a single IAX trunk on each of the interfaces. All 3 interfaces are going to have separate publicly routable IPs, and for this purpose, let's say that because of