similar to: No timezone in Voicemail email?

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid

Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far: [incoming-fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif) exten => s,3,Hangup() exten=>h,1,System(/usr/local/bin/fax2mail --cid-number "0${CALLERIDNUM}" --cid-name "home fax"

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next

Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers Steve

Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Hi All, Which echo cancellation is good between OSLEC and MG2. Dahdi by default use MG2 echo cancellation on channel. If i want to use OSLEC then what should i need to do ? Do i need to recompile dahdi with OSLEC ? -S -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings! We are using cisco 7940 phone with SIP and asterisk. We would like to be able to have phone directories available on the phones that are sourced from active directory. Are their any scripts that can connect to the AD server via LDAP and then create the directory.html file for the phones? Thanks! Liz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? kel

Hello Anyone who have already use/configure Asterisk with CDRTool ? Or maybe can suggest another CDR GUI ? regards. Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090812/e3e9e675/attachment.htm

Hello, How to take the values of channel variables like 'agi_uniqueid' and 'agi_callerid' in agi script. For example #!/bin/bash -x T="$agi_uniqueid" I want to save value of 'agi_uniqueid' channel variable into a variable called 'T' in my script -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just want to pdf and email faxes coming in over pstn on a TDM400P. Outgoing faxes would just go out over pstn, not through asterisk. All the voipinfo , etc, howto's are quite complicated. And most use third party apps like Hylafax. I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm now using 1.4.22, but I'd go to 1.6 if it made this easier. But I've

Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=<providers_IP> and ipaddress=<providers_IP> Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/NUMBER at Provider) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]:

On Thursday 06 July 2006 13:08, samba-request@lists.samba.org wrote: > Message: 29 > Date: Wed, 05 Jul 2006 11:29:23 -0500 > From: Don Meyer <dlmeyer@uiuc.edu> > Subject: Re: [Samba] Print Cost Capture > To: "Samba User's List" <samba@samba.org> [...] > The catch is that it was written around LPRng. ? A couple times now, > I have

Hi Everyone, Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Even though FreePBX Inbound has an option for Alert_INFO but that doesn't work when the call comes into an IVR or Queue. The calls has to go directly to extension for external ringtone to be different. So, I am looking for internal calls

Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. Best Regards; Wassim _________________________________________________________________ Windows Live Spaces is here! It?s easy to create your own personal Web site. http://spaces.live.com/signup.aspx

Hi: When i get an incoming call, i want asterisk to make the caller hear music"musiconhold" instead of ringing,Can any body help me with this? Best regards; Wassim _________________________________________________________________ Get the new Windows Live Messenger! http://get.live.com/messenger/overview