similar to: Changing CALLERIDNUM on the fly

I'm experiencing a strange problem. My inbound calls are hanging up right after Background() message even though response timeout is set to 10 sec. [voicepulseincoming] exten=>_X.,1,Answer exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)

How does one process a return code in Asterisk? Example... exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM}) exten => s,n,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm") If the caller hangs up on the playback command the file remove System statement after it never gets executed. The playback command returns a -1 in this case and logs a warning. The only thing mentioned in the

Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the "011" is the maximum number, because 17 is just not found: OK: 1234567890123456 http://www.google.com/search?q=011XXXXXXXXXXXXX Not OK: 12345678901234567

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

Hi all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten

When a user dials 999 he is always asked for the mailbox and has to enter his mailbox number and password. As I understand this shouldn't happen because the CALLERIDNUM is passed over to VoicemailMain. It's annoying to have to enter the number everytime ... The voice mail configuration is read from MySQL. We are using the CVS version from a few days ago. Extract from extensions.conf:

Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if it is more than 3 digits it rings another i.e. internal calls and external calls. exten => 2222,1,GotoIf($["${CALLERIDNUM}" = "1111"]?5) This will tell it to jump to 5 if callerID if 1111 but how do i

I upgraded from 1.0.9, to 1.2.1. I was using this line exten => s,1,gotoif($[${CALLERIDNUM::3} = 066]?mycity,1:other,1) it selecting calls if callerid begins with some number pattern (from some city) But, it's not working anymore in Asterisk 1.2.1 when I test this with noop(${CALLERIDNUM::3}) I get full callerid, not just first 3 numbers like it use to be on 1.0.9 Why?

I am trying to do something that I see describe in a book and it is not working.... In my sip.conf, I have in my [fxo] context=from-pstn I then have in extensions.conf [from-pstn] exten s,1,answer() exten s,2,playback(blah) etc. It never answers.... but if I do this [from-pstn] exten _x.,1,answer() exten _x.,2,playback(blah) it works. Why does the 's' extension not work here?

Can anyone suggest a source for a free robot dialer or examples? I need to do some non-commercial auto dialing using Asterisk. Simple phone numbers in a file, line by line format. I found one called AstAutoDiaker but I was not able to get it to work and it appears to not be supported - no email response from author. Doug

Hi, I have a PRI line with DID (from 100 to 499) in Italy. Now I'm seeing all calls from same DID 'main' number. Can I set outgoing CallerIdNum to the right extension? Thanks Mimmus

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Hello, In Asterisk 1.4 beta 3, the UPGRADE.txt file says: Variables: * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE}, and ${LANGUAGE} have all been deprecated in favor of their related dialplan functions. You are encouraged to move towards the associated dialplan function, as these

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a "best way." I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

Ok trying this again... is there anyone using the SPA-3000 with * ???? I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a

Being rather new to Asterisk I was wondering what the current status of AEL2 is? I see reference to it back in January but that was many versions ago. Is it in the current code? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

Does it matter if you use upper or lowercase rules - I.E. - "x" vs. "X" or mix them? Not that I would do that as a rule but sometimes you make mistakes! Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

I was wondering if someone might be able to offer a suggestion to me about how I might go about dropping a caller into a context specific to their CID. For example, I would like to be able to dial Asterisk from a specific number (a mobile phone) and have it drop me into a context other then the one that normal callers receive that has more options tailored to things I might want to do. I assume