similar to: Asterisk 1.2.2 - Double Quote on CallerID Causing SIP Problem (7940)

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

Hi all I'm pretty sure someone must have done this before but I couldnt find any trace of it on the web so I thought I would drop a note about how I ended up doing it. I have also posted this info on voip-info. Warning : This is not very elegant and I'm currently trying to write a patch in order to make it better but so far, this the only way I've gotten this to work. Scenario : I

I've got an interesting scenario where transfers while getting an invite seem to break. Here is the scenario: You have a receptionist who has a 6 line phone (in this case, a polycom ip600 - also tested with a Cisco 7960) the receptionist has all six lines available for use (in the case of the cisco I tried registering all lines as one number as well as registering multiple lines and

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

When receiving inbound calls from a Vonage Softphone extension, I'm unable to view/maniupulate calledid data. but it shows up in the CDR records and on called handsets.. any ideas? exten => asda,n,NoOp(callerID is ${CALLERID}) exten => asda,n,NoOp(CallerID is ${CALLERIDNAME}) exten => asda,n,NoOp(CallerID is ${CALLERIDNUM}) -- Executing [asd at pstn-in:2]

I have been struggling with my Asterisk setup for 3 days now and I think I have done well...apart from the small detail that I cannot dial out on my phone (PSTN) line. My setup is: Suse Linux 9.0 1 fxo card connected to a BT(UK) line 1 Cisco ATA186 sip v3.0 with two analogue phones attached to it Asterix CVS-HEAD-05/30/04-06:56:31 with the UK Userid patch applied. Asterisk loads without any

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

Hi, I am running dovecot 0.99-14 on a Fedora Core 2 machine. I had a few questions: 1) I wanted to upgrade to the dovecot-1.0 release. However, I am not sure if that's really required. dovecot-0.99-14 has been running very well for me for quite some time. Is there a real advantage to switching to the latest release. The reason I'm asking this is because: i) I don't have too

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"

Hi, I have two * box. One is box1 and the other is box2. And I have two iax clients A and B. A is registed with box1 and B is registed with box2. If I make a call from A to B using the following method: IAX/[<user>[:<secret>]@]<peer>[:<portno>][/<exten>[@<context>][/<options>]] The CalledID that B got is <user> but not A's CallerID. I

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

I have quite successfully set up the Services button to work on the 7940 running SIP. I have a metric-imperial converter, a foreign exchange rate calculator, a calendar etc available to users. The XML is really fussy though. Simon -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, 1

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

I am just starting out with * so bear with me please. I have tdm400p with 4 fxo modules on it. When I call into the asterisk box from my mobile, I can see the asterisk console picks the call up and routes it to my computer with x-lite. There was no sound coming from either - just silence. I then decided to route it directly to voice mail to see if that would narrow the problem down, but it

Hello Everyone. I usually find my own solutions for problems but this time, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the

I am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------

I have Asterisk deliver all voice mail to users as email attachments. I found by accident that there is a limit of 99 messages in your INBOX in Asterisk. The 100th attempt to record a voice mail causes the system to play your greeting and then never record the 100th message and silently disconnect the caller. So...is it safe to simply use the UNIX find command to delete any files in the