similar to: Voicemail mailbox number passed in connection?

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype should be configured in the zapata.conf file when AT&T is using CUSTOM? My understanding is that

Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten => 202,1,ANSWER() exten => 202,2,PLAYBACK(tt-monkeys) exten => 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [202 at local:1]

Is there a way to allow a user to dial an extension after listening to your voicemail instead of leaving a message? Example would be the big boss is on vacation and changes his out message to say "you can reach my assistant at by dialing 1234 now or leave me a message". Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have the following: Call -> PSTN -> * -> GrandStream 101 1.0.3.81 The GS displays "ohn ro n2600" when the call is past to the GS. If I pass the call to a XTEN client, Caller ID shows up. Any ideas ??

I am having an issue with the CallerID Number not being passed to my phone in the SIP packet. The CallerID Name is passed just fine and displayed on the phone with no issue. I have done a NoOp() in my extension.conf and successfully seen both the CallerID name and number correctly. So that leads me to believe that Asterisk is handeling it correctly. However, when I do a packet capture of the

Has anyone had any experience using a modem through the Asterisk system? I have some technical support personnel that need to use a computer modem to connect to a remote system for troubleshooting. Is there a SIP compliant gateway that will support a modem connection at decent speeds (minimum of 28.8) that anyone knows of? If not, has anyone used a Digium FXS card for this? Thanks

I have an IAX trunk configured between 2 Asterisk servers. Everything is working great except if the caller presses # during the call. If they press # the local PBX comes on and says transferring and tries to transfer to a blank extension. Does anyone know how to turn this off? There is no extension defined for # in the dial plan. Thanks for your thoughts on this.

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

I have come across three problems in the past few days, in spell of heavy R ( version R0.50-a4/Sun Solaris2.5.1 ) use. 1. I was using lwd=2 to get thicker lines on plots for printing, but although the 'lwd' parameter works fine with x11(), the thickened lines do not print with print.plot, or by using postscript() directly. 2. Try the following, plot(1:10, -(1:10))

Hello, There is a problem that i can not figure out how to solve. I got users with 5 digit usernames for sip. Some users has a callerid for outside calls. I have such problems When a user activates (for ex) call forwarding, System creates that entry on database as CFIM/callerid not the username, So this rule works only if a call is made from outside to the callerid. Not the local calls made

hi Guys, As we get ready to start publishing Cloud Images ( or rather images consumable in various virt platforms, including public and private clouds ) - it would be great to have a baseline package manifest worked out. What / how many images should we build. At this time we were thinking of doing : - CentOS-5 32bit minimal - CentOS-6 32bit minimal - CentOS-5 64bit minimal - CentOS-6 64bit

Hello I'm thinking of selling an Asterisk server based on Atcom's IP02 solid-state unit with one FXO and one FXS ports: http://atcom.cn/En_products_IP02.htm By default, this unit based on a 400MHz Blackfin 532 chip only has 64MB RAM and 256MB of NAND flash. Those can be increased to 128MB and 1GB, respectively. Do you think I can install Linux + Asterisk + LAMP (replacing MySQL with

I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a "telemarketer hold" extension which, if I transfer a caller into it, loops around playing music & "please wait" messages, until they give up & hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if

Hello, I've got a new centos 6 system it's i386 and I'm wanting to update it's lamp stack, specifically httpd, php, and mysql. Also, if postfix as an MTA would be available that's a plus. I've checked out the centos wiki and it looks like I have two choices for third party repos for this task. I have enabled rpmforge as of now. The two repos I'm seeing are Les RPM de

Hi all, I set up a Samba server (version 3.0.24) names "Lamp" on a Debian Etch distribution. It provides a "guest" (guest ok = yes, guest only = yes, security = SHARE) access to a "share" directory, which works fine. Indeed, I can access \\lamp\share from a Windows machine (named "Ginger") without issuing a password. I can also ping the Linux machine

Hi, I am trying to install a LAMP stack using MariaDB on CentOS 6.4, 64 bit. I have installed MariaDB by using the MariaDB repository configuraton tool. I installed php by itself. Both MariaDB and php individually test fine. I know that php-mysql has to be installed to complete the LAMP stack installation. From lots of googling I understand that there is a php-mysql and a php-mysqlnd.

I'm using a SNOM 360 with Ver 4.3 software. Asterisk is.... Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff + Head) I've used the wiki info to set up some lines to monitor some internal extensions. When the extension is rung - the lamp comes on, when the call is answered, the lamp goes off.. I was expecting something a little more exciting - like the lamp to flash when the

Hello all, I am very happy to announce that registration for my new course entitled "Ruby on Rails Development" at my college is now open. We are offering this course as part of our LAMP certificate which is designed for working professionals who want to upgrade their skills. I think that it is a sign of Rails'' growing maturity that I was able to propose this and work it