similar to: Asterisks, Sip to Local PRI/PTSN issue

I have the below script that doesn't seem to be working. I don't know if I have something in the script wrong that I am just missing. Or if I don't have the php.ini set correctly for emailing This is the CLI output -- Executing [4099XXXXXX at port3_real:1] Goto("DAHDI/50-1", "newhire,s,1") in new stack -- Goto (newhire,s,1) -- Executing [s at

Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption

Alright I have a SIP phone located off premises with a very annoying issue. Well I say a sip phone it is actually two phones hooked to a Cisco Spa 2102 Link: http://www.cisco.com/en/US/products/ps10026/index.html Each phone being a different line/extension. Alright either line can ALWAYS make outbound calls no issue. The problem is on the Inbound side. I'm completely stumped as

I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this

Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption

There are times when I need to call a client from my cell and I want a recording of the call. I was trying to put into an * a way of doing that. Below is what I'm using in my extensions.conf exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5) exten=> X,2,SayDigits(${num}) exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify) exten=>

Alright I know how to do basic IVR in *. But what I'm working on trying to do now is a survey. I've found very little things out there on google or the archives for how to do surveys with the * ivr. Here is more or less what I'm trying to accomplish 1. Call comes in Plays Greeting 2. Starts Survey 3. Ask Q1, Record the answer (voice responses) repeat this

Other than having stripping out IPs this is what I am receiving for my voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine calls that come in on my PRI. BUT at least from this VOIP source the To field which is my RDNIS information for these calls, doesn't actually fill into ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does

Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if above screenshot from wireshark does not show here is a link for it: http://img402.imageshack.us/i/tempe.jpg/ I've tried a variety of setups in my IAX.conf (they all end up with the same issue, tried just bindaddr=0.0.0.0 with

AGI is executable. =20 Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script =20 I have the

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. Shigley Sent: Wednesday, December 02, 2009 3:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Variable Name needed That wasn't it either. I tried a few other likely fields from

is executable. Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script I have the below

Telco, location, ect?) At X times of day? =20 Ect, ect. =20 It sounds like bleed over, which can be causes by some many things the best place to start is to find a pattern if there is one. =20 James Shigley Monroe Telephone Answering Service =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC Sent: Tuesday, May

Hello, all. In section 4.3.10 of the Asterisk Handbook, there is an example of an LCR/Failover Trunking scenario. I've tried it, and it works, as long as I fail over from something else to ZAP, but I can't get it to "hunt" to the other context if the zapata channel (or group) is used first. Can anyone help? Here is my extensions.conf, and the error message I get.

Ok so a while back I found an example for having a number dial multiple numbers and then whoever answers and confirms gets the call. (don't recall who the example was from, but thank you!) But Now today I've been playing with TTS and STT and came across the BackgroundDetect command. Now If I use this allow it works fine. But when I try and use it with this it never actually detects me

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

In my context I have: exten => _NXXXXXX,1,Set(CHANNEL(musicclass)=default) exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) but the recorded conversation ended in 10min so it = 600sec I was looking in asterisk configuration file for "600" pertaining recording but

In my dial plan I have these two lines: exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) How to add "if" statement to execute these line only after let say 5pm. To record conversation only after 5pm. -- Joseph

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com