similar to: Sending DTMF after a call is set up

I've been asked to recommend a solution for a one-E1-port PSTN gateway supporting SIP. I've never set up a Cisco 5300 or equivalent, but I know they work. I use the Asterisk software in a couple of places and would like to use the E100P. My question is whether anyone out there has any installations using this and what their opinion is about it (does it work? how's the audio quality?

I'm trying out ASTCC. I set the card length to 10, and generated a test card. 10 digits. I set the extensions file to: exten => 9175954700,1,Answer exten => 9175954700,2,DeadAGI(astcc.agi) exten => 9175954700,3,Hangup I dial in and the prompt tells me to enter my 12 digit PIN, not 10 digits. How come it thinks it is 12 digits? I set both the Published number and DID in the Brand

I have a second-hand 7960 which I am attempting to upgrade to use a SIP image. The phone currently has a firmware release which doesn't seem to be listed in Cisco docs - P003AM30. On reboot, it finds the tftp server and requests the firmware image listed in OX79XX.txt correctly, displaying "Upgrading Software" on the screen. It then continues to re-request the same image from the

I need to implement a procedure for creating a 3-way call, similar to what you get from the telephone company. You're in a call, you flash hook to get the switch's attention, you dial the 3rd party, you flash again to create the 3-way call. In the asterisk world, the flash would be replaced with the *+(some key). Is this implemented? How would I configure this? Thanks for any help,

We have been using Samba for many years. The company has just switched from an NT domain to an Active Directory domain. The new server is running Windows Server 2003. We are having trouble configuring our Solaris 8 server so it can join the domain as a server. Just getting Samba to compile and link was interesting enough. This included downloading and compiling a new version of the BerkeleyDB,

I'm looking to use the Digium E1 card in Iran. Does anyone out there have any experience with something like this? What are the odds it will work first time or ever? Bill

I am using "maps". I am running the following code to get this list of all the counties: map('county', plot=FALSE)$names In the output, all counties have first the state listed and then, after a comma, the name of the county. However, county # 2395 (State = south dakota) has no county name. Anyone knows what this county is? Thank you! -- Dimitri Liakhovitski

Hello everyone. This is my first time attempting to use Xen and I am having some problems. I am trying to build a Xen system so that I can use the Ether program for malware analysis (http://ether.gtisc.gatech.edu/source.html). I''ve installed Debian 6, and downloaded the Xen 3.1.0 source. However when I try to compile the source using "make world" I receive the following errors:

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the extension to certain number as I don't know what number I will be dialing. -- #Joseph

I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event --> mobile phone --> software answering machine --> Internet server * Event --> mobile phone --> VOIP --> Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

Hi, How do you transfer (using say blind transfer) a call to an extension that is currently busy on another call? You don't want the call to be transferred to voicemail, it must stay in 'hold' until the extension becomes available, and then immediately ring that phone. Thanks, Thomas

Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. I'm sure someone has done this already. Anyone want

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

I'm getting an INVITE from the Cisco softswitch looking like this (from sip debug trace): Sip read: INVITE sip:9043940358@206.165.120.52;user=phone SIP/2.0 Via: SIP/2.0/UDP sia-ATLCA146.telefyne.com:5060;branch=z9hG4bK_1146_38l5 From: <sip:7278673170@sia-ATLCA146.telefyne.com;user=phone>;tag=1_1146_f153592_732 j To: <sip:9043940358@206.165.120.52;user=phone> Call-ID:

The wiki says this about the MusicOnHold command: "Plays hold music specified by class. If omitted, the default music source for the channel will be used." http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ? I tried adding musiconhold=test to my sip.conf. musiconhold.conf looks like this:

Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like " I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him "wait a sec" and push "Flash" and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to"...

Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler -------------- next

I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar