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We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed

Asterisk seems to regard an unregistered phone to be busy. Is that correct? Is not an unregistered phone unavailable? It is odd to me that if someone dials an unregistered extension, then the dialplan jumps to busy and voicemail kicks in saying that the person is on the phone, when clearly they can't be if the phone hasn't registered. Any way around this?

Try commenting out the line pritrustusercid = yes Or set it to 'no'. That worked for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jens Sent: Friday, January 21, 2005 7:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zaphfc no callerid

For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much.

Hello Asteriskians! I have an Asterisk box with a simple HFC card in it and a bunch of people using the Xlite software to connect. The HFC card is connected to an internal extension on our legacy PBX. So far so good. The Xlite clients can call each other, and the internal extensions on the PBX and the Xlites can call each other, no problem. The problem is when using an Xlite to dial an external

I've been struggling with connection Asterisk to ISDN BRI lines for a while. I have it working with the latest bristuff and compatible Asterisk version: Asterisk 1.0.3-BRIstuffed-0.2.0-RC3a I am using a cheap Centronics ISDN card and the zaphfc drivers. It works but users complain that the sound quality is not good. They have Xlite phones on their desktops. Xlite to Xlite through

What are the advantages in using mISDN over other solutions? If I knew why it was a good idea (like does it have better sound quality than alternatives?) then I would put the time in to test it, and also improve the Wiki. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Thomas H?ger Sent: Friday, January 21,

I guess it is possible to have an HFC-S card and a Digium TE110P card working at the same time? The TE110P will work in E1 mode. I think the zaptel.conf is probably right but the zapata.conf not (I just tacked on another group at the end but I don't really know what I am doing). Can anyone help? ----------------------------------------- My zaptel.conf looks like this: # hfc-s pci a span

I am having problems loading the zaphfc from bristuff and wcte11xp drivers at the same time. If I load zaphfc then all works fine. If I then load wcte11xp, the card using the zaphfc doesn't pick up calls anymore. I am using bristuff 0.2.0-RC5. Anyone else seen this problem, know of a fix, or can tell me what I am doing wrong? Thanks My zaptel.conf: # hfc-s pci a span definition # most of

I have noticed the following: - the PCI ID of the card seems to change over time which means that loading the module does not always recognise the card, only way to reset this is to power cycle the machine - you cannot unload the module once it is loaded, it hangs the machine, which also means if you have automatic shutdown scripts for restarting the machine then the machine will hang on

Anyone have any experience with the Rhino T1 channel bank? It looks very cost effective at around $1300 for 24 lines but I haven't seen it mentioned on the Asterisk list yet.

This comment comes up fairly regularly and is confusing people. Why doesn't it say that it failed so we know? The way it is now it kind of leaves you hanging there and you don't know if the transfer happened or not. And why was it even attempted if it is obvious that transfer is off? (I know it can depend on the remote side.) -----Original Message----- From:

I have the latest bristuff, a zaphfc card for external calls and a PRI card for connecting to a PBX as a channel bank. With a BRI I would expect to be able to have two incoming calls going at the same time, but when I try it, one call connects and the other gives the following console message, a busy tone, and then a hangup: Mar 23 19:18:35 WARNING[5595]: chan_zap.c:7512 zt_pri_error: PRI:

OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem?

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear.

It seems to me silly to have a T1/E1 card to connect to a channel bank when you could just have a 24/30 way FXS card in the slot in the first place. Does such a thing exist? Wouldn't Digium have a lot of customers if they could produce one for say < $1000 retail?