similar to: Caller ID for auto outgoing calls

On 2003-04-26 at 00:42, Jim Gottlieb (that's me) wrote: > [ccmenu] > exten=s,1,Ringing > exten=s,2,Wait,2 > exten=s,3,BackGround(5045) > exten=s,4,Goto,outtrunk|17005554223|1 ; if they just wait > exten=_X,1,Goto,outtrunk|17005554223|1 ; if they press 0-9 > exten=_*,1,Goto,outtrunk|17005554223|1 ; if they press * > exten=_#,1,Goto,outtrunk|17005554223|1 ; if they

For years we've had the following simple context for outgoing calls: [outtrunk] ; match any NANP, and strip leading 1 off exten => _1XXXXXXXXXX,1,StripMSD,1 ; dial outbound on trunk group 1 exten => _XXXXXXXXXX,2,Dial,Zap/g1/${EXTEN} But when I upgraded on Friday to the latest CVSHEAD, this no longer works. If I send 13115552368 to this context, I get a message like pbx.c: Channel

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.

I'm trying to filter calls that don't have a proper ANI. This is what I did: ; only if they a real-looking ANI exten=_1XXXXXX1118/_.N.,1,Newt,1118-config ; Otherwise, send them to the loser partyline exten=_1XXXXXX1118,1,Goto(outtrunk,19096611234,1) This properly deals with null ANIs, but for some reason those with ten zeroes get matched by the first line. I also tried to be a bit more

While R has library TSP to help solve traveling salesperson problems, does anyone know if it has any libraries to help solve multiple traveling salesperson problems? For instance, suppose one is planning school bus routes and one has multiple buses. Thank you for your time. -- View this message in context:

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

I haven't dealt with Centrex for a long time, and one of my customers is being courted heavily by a Sprint salesperson. Am I not correct in assuming that each "line" of Centrex corresponds to an "extension" in the PBX world? This site has 2 POTS lines and 5 extensions, and they told me that for the same thing they're paying right now (~$40/POTS line) they will be

asterisk-16.8.0 Hi I've set up a callback script to retry a number if it's busy, but as I watch the console output asterisk seems to rush 3 or 4 calls at once before waiting the RetryTime of 20 seconds that I've set. The script: -----8<------ CALLERID=$1 EXTENSION=$2 TEMP=`mktemp /tmp/call-XXXXXX`.call cat <<EOF > $TEMP Channel: IAX2/account at

Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing

Hello everyone I'm trying to make asterisk get a call out using the .call system. The setup is A@H 2.6 This is the content of the file is : <<< Channel: Zap/g0/052MYPHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: ext-local Extension: 210 Priority: 1 >>> I'm

I've seen other posts about this problem, but I haven't found a solution. I'm dumping eight call files into the "outgoing" directory at one time. Three of the calls are successful while the other five are lost. Here is the call file: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Note: All

I'm doing some automated calling by putting .call files in the Outgoing folder of Asterisk. I'm concerned this might be a stupid question, but I'm pretty sure I've done my research well and I'm unable to come up with an answer on my own. I want to know: what happens to the .call files after the "MaxRetries" number has been reached? In my experience, they stay in the

Hi all, I am going to be building an Asterisk system to replace the current aging (aged) Nortel Meridian system in a travel agency. There is already a voice T-1 in place and currently there are about 20 extensions in use. I would want to move up to about 25 extensions immediately and about 30-35 within the year. I am going to want IVR and voicemail, plus the ability to ring a group of

I have a PSTN termination provider "foo" which will accept standard U.S. calls in the form 1<10 digit ph#>. I have an outbound route named "foo", with dial pattern "5|.", with the only entry in trunk sequence being "IAX2/foo". I have an X-lite local extension, on which I can dial 51<10 digit ph#>, and asterisk will call out over foo and the

Hi, I have controller code that needs to be shared between multiple different user type views and want to know what is considered ''best practice'' or what other people are doing out there. Example: I have admin users, salesperson users, and possibly another type of user. They all need code to add/edit/delete a property - and other abilities. The code would be identical(in

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? Please help me, where can I add SipAddHeader() in

I have created a call file as shown in the files below. The number is dialled and connected (i.e. the call is placed to the PSTN) but it is immediately disconnected and I get the following message on the console: Starting Zap/3-1 at from-internal-custom,s,1 failed so falling back to exten 's' Extensions.conf [from-internal-custom] exten => s,1,Wait(20) exten => s,2,SendDTMF(1) etc

> you should mv the file (and in the same filesystem, so 'rename' is used) > You might want to chmod or even chown the file first as well. I wrote a little script that does all of this before the .call file is mv'd into the outgoing directory: cp /tmp/test3.call /tmp/test1.call chmod 666 /tmp/test1.call chgrp asterisk /tmp/test1.call chown asterisk /tmp/test1.call mv