similar to: Asterisk and Linejacks

Hi all, Does anyone know how to get the channel ID on the other side of the call? For example: When SIP/50 calls SIP/21, and the call is answered by SIP/21 I get: SIP/21-6735 answered SIP/50-b456 ${CHANNEL} will show me SIP/50-b456. Is there a parameter or a workaround to get the SIP/21-6735 part? Thanks. Assaf Benharoosh -------------- next part -------------- An HTML attachment was

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

Hi folks, First off all, I'm new to asterisk. I'd like to make a basic installation of * with a Linejack, as recommended in this mailing list, I downloaded and setup successfuly the pwlib, openh323 lib, asterisk and chan_323. I have made little changes to default configuration files packed with asterisk and I managed to call with the telephone plugged in my Linejack a friend

To whom it may concern, When dialing out an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This is not due to a problem on your Asterisk or your line- the bad sound effect occurs in VoicePulse. (just spend lots of time finding that out) Assaf Benharoosh MCP, MCSA, MCSE ab@franticllc.com <mailto:ab@franticllc.com> Frantic, LLC. -------------- next part

Hi All, Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)? The card works perfectly with virtually anything else but asterisk. Maybe the CVS versions have some work on it? Cheers, -Z -- __________________________________________________________ Sign-up for your own FREE Personalized E-mail at Mail.com

Hi All, Did anyone came across a 411 XML service I can feed to the "service" button with XML? Some other feed I can manipulate to XML query? Assaf Benharoosh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/dbf552ac/attachment.htm

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

I've done a quick search through the old mailing list archives to see if I could dial an outside line through * using a quicknet linejack setup as an FXO. Is it posible to do so? It appears from the old mailinglist from marko.net that this not possible? Thanks. Bradley Greep

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Hello, I have ISA card LineJack. I could not find any information if this card can work as fxo with Asterisk. If it can work, can somebody point me how to install it on my Asterisk box. Or maybe there is some documentation about it how to install LineJack. I will be very thankful for any help. Regards Bartosz Jozwiak ------------------------------------------------- This mail sent through IMP:

Hi, New to the list. Just wanted to know if someone managed to get LineJack to make outgoing calls to the PSTN line in Asterisk. Any dev on this? Cheers, -P -- __________________________________________________________ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup

Hi, I am trying to get a prototype working based on Asterisk and Quicknet cards. I currently have to systems set up each with a LineJack card. I have the systems working, but can't get the voicemail demo to work properly. Messages to the user telephone set from the voicemail system are clear, but recordings left through the phone are distorted. Sounds like the audio has lots of echo and

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Hi, I am successful in getting incoming calls for my linejack .... But, I am getting an error cannot create channel of type Phone.... Does linejack do outgoing calls in asterisk ? Thanks, PTA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031011/79060c3b/attachment.htm

I have a Quicknet Linejack. Actually, I want to make a standalone(no internet connections) call to my asterisk box running GNU/Linux. When I call the system it says: -- Executing BackGround("Phone/phone0", "demo-congrats") in new stack NOTICE[14349]: File channel.c, Line 1212 (ast_set_write_format): Unable to find a path from 2 to 1 WARNING[14349]: File file.c, Line 553

My thoughts on this after reading Steven's very politically worded reply is that IMO your best bet would be to go with RedHat, I am not going to go into details about the if's, when's, why's, and but's.. I am running Asterisk quite well on RedHat 9 and if you like I have created an install guide for setting up an Asterisk box on RedHat 9 which I can send to you if you are

Hi, I'm having problems with setting up my ISA LineJack card on Linux machine... I've done everithing according to documentation available, by compiling the new ixj driver (v1.2.1), loading it, adding device node /dev/phone0 with major number 100 and minor number 0, adding aliases into the /etc/modules.conf: alias char-major-100 phonedev alias char-major-100-0 ixj and running depmod

Hello All... I have searched in the archive and also followed Zara's instruction on getting incoming calls to work with Asterisk...but I still can't get Asterisk to answer incoming call on Linejack's LINE port. I attached a phone set to the PHONE port, and telco line to the LINE port on the Linejack(ISA) card. I have downloaded, compiled and installed the newest driver for

For some reason I can't get the Dial command 'C' flag to work. The calls are recorded in the CDR with the 'C' on. Does anyone have an idea? extensions.conf: exten => 114,1,Dial(SIP/114,,C) It shows in the lastapp: cdr: | 2004-10-05 13:16:02 | "112" | | 114 | intern | SIP/112-3fb6 | SIP/114-0e7a | Dial | SIP/114||c | 6 |

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this