search for: answeringserv

Displaying 15 results from an estimated 15 matches for "answeringserv".

2009 Apr 23
3
AGI PHP script
...lerid; $agi_calleridname; $agi_extension; $agi_uniqueid; $UNIQUEID = $agi_uniqueid; $CALLERID = $agi_callerid; $EXTEN = $agi_extension; $attachment = "/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav "; $from = "... at xxx.com"; $to ="jas at answeringserv.com "; $subject="New Applicant"; $headers = "From: $from"; $message ="$UNIQUEID , $CALLERID , $EXTEN , $attachment"; mail($to,$subject,$message,$headers); ?> So is it anything obviously wrong with the script I'm missing? Besides something no...
2009 Dec 28
2
SIP Issue
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
2009 Jan 16
0
No subject
...callerid; $agi_calleridname; $agi_extension; $agi_uniqueid; $UNIQUEID = $agi_uniqueid; $CALLERID = $agi_callerid; $EXTEN = $agi_extension; $attachment = "/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav"; $from = ". at xxx.com"; $to ="jas at answeringserv.com "; $subject="New Applicant"; $headers = "From: $from"; $message ="$UNIQUEID , $CALLERID , $EXTEN , $attachment"; mail($to,$subject,$message,$headers); ?> So is it anything obviously wrong with the script I'm missing? Besides something no...
2009 Jan 16
0
No subject
...alleridname; $agi_extension; $agi_uniqueid; $UNIQUEID =3D $agi_uniqueid; $CALLERID =3D $agi_callerid; $EXTEN =3D $agi_extension; $attachment =3D "/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav "; $from =3D "... at xxx.com";=20 $to =3D"jas at answeringserv.com "; $subject=3D"New Applicant"; $headers =3D "From: $from"; $message =3D"$UNIQUEID , $CALLERID , $EXTEN , $attachment"; mail($to,$subject,$message,$headers); ?> =20 =20 So is it anything obviously wrong with the script I'm missing? =20 Besides...
2010 Feb 24
0
Question
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
2010 Jul 19
1
Asterisk Queue + Caller ID issue
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
2010 Jul 16
1
(no subject)
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
2009 Dec 04
1
IAX2 Port issue
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
2009 Dec 02
0
FW: Variable Name needed
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
2009 Apr 10
2
IVR Survey
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
2009 Dec 02
2
Variable Name needed
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 ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does
2009 May 04
3
AGI PHP
I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this
2010 Feb 02
2
Semi-Transfer
There are times when I need to call a client from my cell and I want a recording of the call. I was trying to put into an * a way of doing that. Below is what I'm using in my extensions.conf exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5) exten=> X,2,SayDigits(${num}) exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify) exten=>
2009 Aug 21
1
Queue Question
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 =>
2009 Jun 17
3
Asterisks, Sip to Local PRI/PTSN issue
Alright I've been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn't. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these == Using SIP RTP