similar to: GotoIf() on Auto-Attendant

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the

Hello all, Where can I find a list of description for each sound files provided by the asterisk-sounds-main Debian package? You can find the contents of my /usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679. Thank you in advance. GNUbie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]:

Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you point me to the right documentation on how to do this or probably hints on how to do this? Thank you in advance. GNUbie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am planning to setup a MeetMe conference functionality on Asterisk-1.4.13without having a Zaptel card. All users will be calling through SIP only. AFAIK, the said application needs a timer which makes use of the ztdummy module. I have basically two (2) problems I am encountering here that [1] I can't load the ztdummy.ko module and [2] Asterisk don't run when running it

I never heard anything on the AMP list, so figured maybe someone here might be able to help me sort this one out.. I was making some updates to my attendant config, which is really very basic, and now incoming call processing stopped. Not sure exactly what the heck happened, but figured maybe someone could help me with a clue as to what broke. Now incoming calls are not being answered at

I'm having trouble finding documentation for the auto attendant does anyone have an idea where there might be some???

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER

I have added two extentsions. I am try to test connecting X-lite to the server. I have two extension one 1000 with password 1234 and one 2000 with password 2000. I have the softphone on the same network so I do not have to worry about ports being open. So I have in the properties of Account Display Name: Andrew Username :1000 Password: 1234 Authorization name :1000 Domain:192.168.3.128

Is there a way to combine these lines into one? exten => s,2,GotoIf($["${CALLERIDNUM:0:3}" = "800"]?s|108) exten => s,3,GotoIf($["${CALLERIDNUM:0:3}" = "866"]?s|108) exten => s,4,GotoIf($["${CALLERIDNUM:0:3}" = "877"]?s|108) exten => s,5,GotoIf($["${CALLERIDNUM:0:3}" = "888"]?s|108) Thanks --John

All, I'm trying to hear the asterisk's auto attendant in its default configuration. According to VoIP Hacks in Chapter 4, I found the following excerpt after successfully configuring my SIP IP Phone (Cisco 7960G): In its default configuration, Asterisk has an auto-attendant that can route calls. To try it out, take the IP phone off the hook and dial 2. Then dial the BudgeTone's Send

Hello all, Okay when you are done laughing at the simplicity of this question could someone show me please what I have wrong in the following statement? GoToIf($[${numdial} != [1-9] ]?15:3); What this is supposed to do is if numdial is not a single digit from 1 to 9 inclusive goto 15, if it is a singledigit from 1 to 9 inclusive goto 3. Should be pretty simple but not working for me, always

Hi Gang, I'm testing 1.8.0 on one of my machines and this snippet "chokes" on line 7 (works fine with 1.4.30) [tb-account-balance] exten => s,1,Set(BALCOUNT=0) exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} )) exten => s,n(runagi),Set(TEST_RETURN="NONE") exten =>

(Both on Asterisk 1.2 and 1.4) I was struggling to find out why my CDR was recording dst = h after a call hangup. It was working fine until I added a GotoIf statement before ResetCDR to calculate some value for userfield column. Today I tested and found out that if ResetCDR is put after GotoIf (or after if in AEL), it doesn't record correct value in dst column, and isntead puts 'h'

I'm testing GotoIf($["${CALLERID(num) but I'm missing something as it is not working: [office-open] exten => s,1,Wait(1) exten => s,2,Answer() ; for Caller ID is 471-5665, always signal congestion: exten => s,3,GotoIf($["${CALLERID(num)}" = "4715665"]?4:6) exten => s,4,Playtones(congestion) exten => s,5,Congestion(5) exten =>

hi everyone. can anyone provide me concrete examples on how to use the GotoIf application? can't figure out how to use it in my dialplan coz im having errors....thanks! : ) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was

Hello! this is a short one: in a gotoif-statement i would like to match a variable to a number, where the number could have digits from 2-6. asterisk only seems to be capable to match such a digit-range when used in the extension, but not in a regexp, at least the following query doesn't work: exten => _X.,1,GotoIf($[${EXTEN} : 234[2-6]]?jump:) obviously asterisk has a problem with

Someone will probably tell you with more certainty, but (you don't say but I assume you are talking about FXO) the Caller ID normally comes in between the first and seconds rings, I think you can tell asterisk not to get the CID but if you don't, it waits for it. Also, I remember reading in a modem manual something about the number of rings you had to let your modem ring before doing an