similar to: Correct setup for directing already ringing calls to newly available phones

Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and

Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was

Greetings. I wonder what is the best way in your opinion to share real-time extension state with applications outside of asterisk? What I'm after is the best way to have Asterisk update a central repository with the state of each extension configured in the local Asterisk setup. To try and explain what I am trying to achieve, Imagine for example if asterisk would call a url like this:

Hi. I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time from a remote connection coming in on TCP. Basically what I have is a Windows application that is used to process incoming, outgoing and missed call records putting them into a database for some analysing etc. This app can connect to a TCP server and read from this connection the CDR's as they are

Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you

I have a call coming in from Asterisk-A going to Asterisk-B where it's determined that the called party is in fact yet another number in Asterisk-A so a new call is created from B to A and the two calls bridged (by Asterisk) at Asterisk-B. Originating Caller ==> Asterisk-A ==> Asterisk-B ==> Asterisk-A Now, what happens is that in my case both A and B are on the same network

Greetings. Just updated from 1.4.22 to 1.8. Minor changes in dialplan and things work ok. Except for one thing. I have a call to MixMonitor. This is implementing a dictaphone kind of app. With forwarding recordings to email and storing them on the server. The process works so that we dial into Asterisk and answer the phone, initiate MixMontior and WaitExten until recording finishes. Problem is

I've looked around and not found details on the expected frequency range the Speex codec can be expected to carry. Is there any documentation available or a table of some sort that has been compiled which would give an indication of the frequency range based on the various compression options in speex? Best regards, Baldvin Hansson Reykjavik, Iceland baldvin@baldvin.com -------------- next

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Is it possible/supported to install two or more B410P Digium cards in one computer (single Asterisk installation)? 2) Do they need to be hard-wired together with a PCM cable like I've seen explained in some beronet manuals (although that was specifically geared towards their cards, I must say)? Thank you for your time and

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Not being able to build mISDN on Ubuntu using "make b410p" I have used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect this version of mISDN to work ok with these cards? Or is there a way to build using "make b410P" on Ubuntu? (make force does not help at all) 2) In some of our installations

Hello, I want that after client and queue member call would be established, cmd queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This is my example of ael : context QUEUE { _X. => { Ringing(); Wait(4); Answer(); Queue(${Queue},wr,,,60,,,check-record); Hangup(); }; }; macro check-record() {

https://bugzilla.samba.org/show_bug.cgi?id=2216 ------- Additional Comments From baldvin@angel.elte.hu 2005-01-06 10:33 ------- I tried the cvs version: it works OK. However, 2.6.3 reproducably hangs. In the NEWS: - Fixed a potential hang when verbosity is high, the client side is the sender, and the file-list is large. OK, maybe this is it. I checked cvs log, and cvs

https://bugzilla.samba.org/show_bug.cgi?id=2218 Summary: inplace-if-low-disk Product: rsync Version: 2.6.3 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: baldvin@angel.elte.hu QAContact:

Dear all. Questions regarding VoIP implementation and the use of the Speex jitter buffer, if I may: Am I right in my understanding that the Speex jitter buffer implementation is used only on the receiving end of a network VoIP stream? 1) The sender would sample+encode+timestamp packets/frames of speex data and send via UDP to receiver. UDP packet would be constructed as: [TIMESTAMP][Speex

Hi, I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. Is there any work around? On voip-info I read

[The following is perhaps a long question and even off-speex topic,] [but if anyone can at least point me in the right direction for ] [alternate sources of information, I'd really appreciate it. ] When speex_jitter_get(...) is called and there is no buffer/data to return, would I not want to know that there is no true data to play? If I turn around and queue 20ms of silence to play

Hi. Anyhone have any experience with trunking between Ericsson MD110 and Asterisk using H.323? I've tried both ooh323 and the /channels/h323 one in version 1.4.4 and 1.4.0 of Asterisk. ooh323 does not manage to establish the call (starts to ring but then disconnection when answering the call on the Asterisk end) but using the channels/h323 driver I can get the call established from

https://bugzilla.samba.org/show_bug.cgi?id=2216 Summary: remote dies, local hangs when disk full Product: rsync Version: 2.6.3 Platform: All OS/Version: Linux Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: baldvin@angel.elte.hu

Hi. I have two B410P ISDN BRI cards in one machine running Asterisk on Ubuntu 7.04. One card connects to the PSTN network and is therefore in TE mode on all four ports and the other card is in NT mode and connects to a PBX. The Asterisk is used to remap features, callerid's and more from the PSTN to the PBX. 1) Is there any special care I need to take regarding the configuration for