similar to: Remote * Using IAX

I'm using happily this cheap phones, but I still have a little problem. Configuring the phone is extremely easy on * and I've a couple of them perfectly working, except when i try to call some toll-free number (in italy 800xxxxxxx ). If the number called is an IVR system, often with GrandStream (but also with Cisco 7905.h323) it's impossible to make the menu choices via the Dialpad.

Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a

Greetings. Today I've installed a fresh new E100P on a EuroISDN PRI. It seems to work well, accepting calls, but, when I start *, I have the screen flooded with this message: PRI got event: 8 (or 6) If i look into /var/log/asterisk/messages i've this Error: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Read on 35 failed: Unknown error 500 Repeated indefinitely. Also, i've a

I've to project and build a fresh new box with * on. Basically, i'll have this situation: The office is connected to the phone carrier with a PRI. I need to let users continue to use their analog phones in a office, and an IP-phone based solution on the remote office, that will call using outside using the PRI on the first one. PRI---> Asterisk ---> Analog phones |

After a long working evening yesterday, now my * box place and receive calls with H323,SIP and ISDN line. Calling from the office to an outside line, happens: - If I call a mobile number and the called answers, all goes ok - If I call a number at home/office, and it's answered , all goes ok - If I call a toll-free number with an IVR system, nothing happens: it continues to ring indefinitely

Is there any way to run asterisk without a fxo card? I am looking only run SIP and a single fxs card.

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously

Hello all, i'm experiencing a list of unpredictables hangup on SIP phones using a PRI E100P Card. All i can see in logs is " WARNING[81931]: File chan_zap.c, Line 5402 (zt_pri_error): PRI: Read on 37 failed: Unknown error 500" I receive a lot of these errors in asterisk/messages. It doesn't seem to be strictly linked to hangups, since i have dozen of these messages per

I''m developing an rails app. The SVN repository resides on a (Win 2k) server on my company''s local intranet. The repository is accessed via http (Apache + mod_dav_svn). From this machine, it is not a problem to get out onto the Internet. However, coming in from the Internet is not possible (at least, not without a VPN connection). Given this, is it possible to use

The hack below is for OpenH323, not Asterisk. This is not an Asterisk problem AFAICT. I am posting it here so that any other Asterisk user with a similar problem might benefit from it. I may or may not post it to an OpenH323 list, but since both variants of the H.323 channel in Asterisk use non-current OpenH323 versions, it may not be of any benefit to anyone anytime soon if I went that route!

Hello, This morning I just started to have this problem calling from a SIP phone to a regular phone, using one of the 4 BRI cards (passive) I've in my * box. It calls regularly, but somewhere after 8-10 secs, it random hangups, or it hangups immediately after a hold, and so. I've looked into /var/log/asterisk/messages, and this is the output corresponding to the hangup: ##### Jul 30

Hello all, I'm becoming mad in trying to solve that issue. If I make a call from any of the phone here (I have some Grandstream and a couple of Snom105 - quite one of the best phones i've ever seen, this last one), to an outside IVR system, if i try to send dtmf to choose one of the IVR options, i notice in the /log/asterisk/messages this line: WARNING[43028]: Discarding too big frame

Hello all, I was playing with the zaphfc driver and i had these issues: I've tried to configure multiple spans (in TE mode, not NT) but it always give me errore "No such Device (6)"... I've 3 HFC-S based cards in the machine, but it seems to load only the first one. If i try to load only the first card, asterisk starts correctly, but if i try to place a call it gives the error

Hello, I did this post a long time ago but never solved the problem, so i'm trying again after something like 10 months, hopefully i'll find someone that found a solution ;-) When i call an external number that sends audio before call has been answered (like some PBX of public offices do here in italy), strange things happen: I'm using chan_capi, with Early B3 active, i can listen

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Hello all, I've just finished to install chan_capi with 3 AVM Fritz PCI cards. It correctly loads the 3 drivers, and * starts without errors. immediately after * start, audio quality is really fine, but, after the first incoming call, all incoming audio is broken, trembling and stuttering.

Hello, i have asterisk with 2 internal isdn cards - handled by isdn4linux and i need to setup whol system like this route some call beggins with 0 or 00 - long distance through first card, route calls to mobile network via second card ( tehere is isdn gsm gateway connected).how i can do this using only isdn4linux (/dev/ttyi) ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with