similar to: BudgeTone 100 Calling Problems

I need to configure an * box to connect to a primary * box which is attached to a PRI in order to make calls using both the same E1 connection. I know the best solution is to use IAX (altough i could connect the IP phones on the remote site directly to the main * box that is VPNed with the remote one). I've some doubt on how to config IAX to work in this situation. I think that: On the

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

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

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

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 would like to know if anybody has test various usb phones with an usb hub conected to an asterisk server. The idea is to build a public telephony site with one or 2 analog cards 4 or 8 FXO, least cost routing and billing and usb handsets with dialpad for users, so I really don't need a graphical interface for the soft phone ( the users just dial from dialpad). Any has done this

Hey - I know there's some other people out there that have the 9133i ... has anyone gotten the DTMF tones to work after the far side picks up? I didn't have any problems out of the box with my SPA-841 phones... the aastra has been nicer so far, but I can't seem to get it to dial the touch tones after an auto-answer device picks up on the far side... I googled, to no avail. -Karl

This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of