similar to: auto dial out multiple destinations

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

Hey, I've coded an OGG player for Win32 (it uses AL for playback so it's portable to Linux/Mac), but every time the program gets to the 'ov_open()' function, the app completely freezes, and I have to use the task-manager to kill it. I am supplying it with a valid file handle that was just opened (FILE*) and the vorbis file is also a pointer that is not in use (set to null). Any

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a

Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -

Hi, How can I match an extension "ending with 3" (just an example but applicable to any other digit, including * or #)? exten => _ZX.3,n,... exten => _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: "$")? Thanks, Vieri

Hi, I'd like to implement the following: Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent to a custom IVR that says something like "extension $EXTEN is $DIALSTATUS. Please try again later or dial $CODE now to notify you as soon as $EXTN is available.". So the "notification" part is what I'm trying to figure out. The extensions are SIP (but

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hi, I have two types of UPS devices and one of them (APC) gave me a false "low battery" alert. I can't reproduce the problem... Also, there was nothing apparently wrong with the UPS at the time of the false alert. All I could do was grab the driver values within a minute after the "low battery" notification. These are the values: battery.runtime.low : 120

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal?

When I compile dahdi I see these warnings: WARNING: "kasprintf" [dahdi-linux-2.5.0.2/drivers/dahdi/wctdm24xxp/wctdm24xxp.ko] undefined! WARNING: "kasprintf" [dahdi-linux-2.5.0.2/drivers/dahdi/dahdi.ko] undefined! And "modinfo dahdi" shows that the driver was built for a 2.6.17 kernel, SMP mod_unload 586 4KSTACKS gcc-4.1 If I "modprobe -a dahdi", I get

you can do this with MeetMe, but you don't have to. you can also use Parking, which makes things a little simpler. in either case, the strategy is going to be something like: 1. Record the soundfile 2. Park the inbound caller 3. Use a .call file or the manager interface to initiate an outbound call to the mobile phone 4. play soundfile and prompt the mobile phone user to accept/reject the

Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [<f8e248b4>] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 00000130 ebx: 00000000 ecx: 00220028 edx: 00000978 esi: 346e5802 edi: 00000000 ebp: c3b45500 esp:

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] <all.eforums at gmail.com> wrote: > Hello All, > > Probably a silly question, but we're wondering if people have had any > experience and have data to demonstrate if the performance of the Asterisk > system might suffer in terms of latency etc. if we're to have it retrieve > sound files from a database using odbc as

I already have ilbc installed on my system. The files are: /usr/include/ilbc/iLBC_decode.h /usr/include/ilbc/iLBC_define.h /usr/include/ilbc/iLBC_encode.h /usr/lib/libilbc.a /usr/lib/libilbc.la /usr/lib/libilbc.so -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0 -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0.0.0 However, if I do a "make" in asterisk-1.4.19, it will not detect that libilbc.a

Hi, My initial setup for receiving faxes worked as follows: fax call arrives on ISDN BRI connected to a BOSCH PBX, signal sent to ALCATEL PBX via PRI QSIG then finally sent to ASTERISK via PRI EUROISDN. The Asterisk server then forwarded the call to a iaxmodem and HylaFax received the data. All worked fine. Now I got rid of both BOSCH and ALCATEL in the "fax path" and it's as

Are shorewall-shell and shorewall-common required at compile time even if one only wishes to use shorewall-perl (4.0.9)? ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ