similar to: Problem Asterisk: can't make call but can receive calls

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

Hello I used to have an extension like this which worked fine with asterisk 1.0.7 I first dial to see if an IAX phone is present, if not I would try on SIP instead exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout exten=s,2,Goto(s-${DIALSTATUS},1) ; Default action exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301

Hello Steve. On 17/05/2005, at 10:54 PM, Steve Underwood wrote: > When a fax is received the header you see is part of the image. As > such, it ends up in the TIFF file as part of the image. It is not > available as text anywhere. The only way to make it available as > text would be to implement OCR. That is complex and messy, and I > have no intention to try. The only

Hello I upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10) And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one. In the UPGRADING.txt in Asterisk it says: * The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key)

Dear all I have a Asterisk server (version 1.0.7) running with a TE110P card on a 3Ghz P4 machine (running nothing else but asterisk) So far so good. Today I was trying to receive faxes using spandsp fax client. I used spandsp v0.0.2pre18 (which compiled against libtiff 3.5.7) It compiled fine, installed fine, run fine. But when sending a fax (from a Brother fax machine) the image received

Dear all. I recently subscribed to a VoIP provider through IAX. The require to connect with either G729 or GSM, I chose G729 based on their recommendation. The service works very well, however ... people mentions how distorted our voice sounds. We have plenty of bandwidth available so I don't think it comes from our side. What it means is that it comes from two things: 1-G729 gives bad

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I'm the one who posted a message about the fact that nobody answer anymore to questions asked. I posted two days ago a problem I was facing that calls made over the Internet to my Asterisk gateway would hang-up after just 5s (no NAT were involved) Answer I got was: it's your config Well, it wasn't (as I was expecting). I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all. I'm new to this so please forgive my ignorance if I missed something obvious. I've set-up asterisk 0.9_1 on a FreeBSD 4.10 server (I know it's not linux but that's all we have available at that stage). After some struggle to understand how everything works, I set up some SIP accounts for test purposes. I can log in,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all. As I couldn't get to compile and run Asterisk 1.0RC1 on my default RedHat 9 I thought it was about time to upgrade to Fedora Core 2. Well, it was too late to realize the kernel 2.6 wasn't supported by Asterisk *officially* anyway. Here is what I did to get asterisk and zaptel to work under Fedora Core 2: I posted it on the wiki

Hello I currently run a few samba servers one being used as a PDC. Today I added a user to the domain and for some reason I can not get it to connect to any of the shares but "home" on the file server. % smbclient -U gregi //server3/public Password: Domain=[HYDRIX-MALVERN] OS=[Unix] Server=[Samba 3.0.28] tree connect failed: NT_STATUS_ACCESS_DENIED However I can connect with : $

Hello I am running the following setup in order to make VoIP calls at home. Home Phone <-> SPA3000 <-> Asterisk Home <- IAX2 over Internet <-> Asterisk Office The voice mail for Home Phone is hosted on the Asterisk Office machine. I wanted to have a way to check the status of my voicemail on my home phone directly. In order to do so, I've patched both Asterisk Home and

Hi there. I have a FreeBSD server running Samba 3.3, connected to a domain who's PDC is a MacOS 10.6 server running Samba 3.0.28 (ancient I know). Working all fine, except for one thing I find annoying. MacOS server has a concept of username alias. You can have as many aliases as you want, using any of those aliases are the same as using the primary one. It's rather well implemented in

I'm looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050515/d66e6335/attachment.htm

We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest. ftp://ftp.digium.com/pub/asterisk Enjoy the code. Special thanks to all the bug marshals and contributers and to everyone who has supported Asterisk

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the

> On 02/06/2011 06:30 PM, Jean-Yves Avenard wrote: >> >> Hi >> >> On 7 February 2011 09:32, Eric Shubert<ejs at shubes.net> ?wrote: >>> >>> Terry was asking about roaming profiles specifically, not a stock pdc. >>> >> >> My bad ; I had missed that part. >> >> Having roaming profile these days is something you get

>On Sun, Jul 20, 2008 at 11:39:15PM +0800, chenyun_zhou wrote: >> >> As we know, if we change the permission of windows' file >> share's permission, the client will determine this. But >> if we change the file share's permission in the smb.conf >> , the client can not determine this if it aready >> connected to the server. How can we let

Hi, I was wondering if it is possible to setup with Asterisk a Cisco 7960 to use extension mobility / roaming. Meaning that a user logs into a phone and his profile moves with him / her. I have a network of ~75 Cisco 7960 phones, running SIP 7.5 distributed across 2 asterisk servers in 2 cities. I would like to enable the users to have the capability to move from one phone to another one

I saw somewhere that the last kernel to work properly with the zaptel drivers when using data over it was 2.4.20. Has this been since fixed to work with newer kernels? -Mike