similar to: Call transfer problems, SOLVED

I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

Same thing here. I had this problem awhile ago and made this workaround. Going to another trunk does not work because they are answering and not sending a error code. If you are using AAH code then this waits 10 seconds on your Voip then times out and goes to PSTN. You can modify for your needs The pertinent line is 14 in macro-dialout-trunk I am going to clean it up and repost under my

Hello, I've got an Asterisk system with 3 SIP trunks configured. Each SIP trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound call routing (via AMP 1.10.007a) uses the 3 trunks in descending order, all set with max channels to 4. Unfortunately, when the first trunk reports a "480 Service Unavailable" (all ports in use), Asterisk reports congestion without

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Thank you (for spamming) - it was the clue I needed to push this through. Sorry it took me a while (and a google :-) ) to realize you'd addressed my initial query - basically, my loss. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Becker Sent: Wednesday, August 24, 2005 00:36 To: Asterisk Users

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote: > Hi! > I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide > a conference bridge for an existing Avaya PBX. I have no control over the > Avaya system, but I am able to speak with the admin in charge when I need > stuff done. I am running all this in a VirtualBox

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

Quick question...Im authenticate all exten except this one(2006). If I call from ext 2006 I still have to authenticate. If I call form any other ext I have to authenticate. Any suggestions? Thanks extex => s,1,GotoIf($[${EXTEN} = "2006"]?3) exten => s,2,Authenticate(731) exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) exten =>

Hello, So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly); exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}) If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing.. Here's a paste of a few things out of the two files that I

Dear R experts, I got help to build a loop but there is a bug inside it that causes one part of the mechanism to fail. It should grow once, but if keep growing on rows where $ti_all is not NA. Here is a wall of code that very crudely demonstrates the problem, there is a couple of dim() outputs at the end where you can see how it the second time around keeps adds (2) rows, but this does not

I'm running Samba 4.0.9 as a domain controller in a Windows 2008 R2 functional level single domain forest with a Windows 2012 domain controller. I am using Bind 9.9.3-P2 as the DNS backend with dlz_bind9_9.so driver. When I add my Samba DC to the DNS Manager snap-in, my main domain.com zone has a red X and I cannot expand it. The _msdcs.domain.com zone seems to be ok. Also, one of my reverse

Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting assembly? I can get one to work, but when I install two, one always fails. I've tried all my cards solo in the enclosure, on each side, and they all work properly when only 1 is installed, however, when I install two, one of them will come up, but the other always fails. Anyone know what might be causing this?

30 says it's view only in the docs & I can't seem to change it, any other options? > Option 30 allows to set Module Shelf Address/ID.

Following up with some more tests... I just left a message in a voicemail box, and got the same behaviour (i.e. it says its starting to play ('First Message') then it goes to the VM ending menu (i.e. press 7 to delete) without playing the message. On this particular test.. I went ahead & left a 2nd VM in the inbox, then when I went to check them, they would both play properly... The

Hey all, been noticing some oddness on a new AAH install... occasionally an incoming zap line with automatically connect with an outgoing extension, even though the incoming line hasn't specified what extension it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's trying to call out from inside the office & are automatically connected with an incoming line.

Hey all again, I'm wrestling with echo problems on our sip extensions. I've set these items in zapata.conf but tweaking these values doesn't seem to make much difference echocancel=yes echocancelwhenbridged=yes echotraining=2500 rxgain=8.0 txgain=1.0 are there other settings that can help me tame this beast? Been searching but not turning up anything that'll work here. Thanks

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

Dear All I have jus migrated to centos 5 and jus wondering why n where this problem could be server A which is running for the past 2 years used as my primary dns and mail server redhat linux 8 sendmail ver-8.12.5-7 bind ver -9.2.1-9 since the server is old and is prone to all security problems i have a new machine which i have installed the following and will replace my existing redhat dns +