similar to: howto on ISDN HFC cards with AAH v1.1

Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these

Hi All, It's broken !! (drat) Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] =>

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? >> Maybe this is rather a hardware question,

Hi All, Long time no chat ;-) Asterisk 1.0.9 (sometimes) won't authenticate IAX phones after re-boot of SuSE 9.3 box I've traced the problem to be with the firewall and the fact that I have 2 NICs in the box. Now that I have opened port 4569 on both interfaces, asterisk seems happy *but* does anyone know how to force SuSE 9.3 to always bring up a specific NIC before the otherone? On

Hi, I've been following a similar and current thread about name resolution from LINUX side. I have exactly the opposite problem (running ubuntu 7.10 on a network with a mixture ubuntu 6.06, WinXP and WinVista boxes) The winboxes cannot see or mount this 7.10 box I can mount WinPC shares on this 7.10 box. I was wondering about firewalls, but I can ping this 7.10 from the WinBox by IP address

Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable.

Hi, Can any point me in the right direction to solve this one? I get a: "this account is not authorized to login from this station" message when I try to "Map Network Drive" from NT4server SP4 to my RH Linux 6.0 Samba 2.0.3, yet I can open a telnet session from the NT machine to Linux using this user, and, in a linux xterm window I can list the samba services using

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

Hi, I'm trying to set up two ACT SIP/IAX capable phones to communicate with each other on the same internal network, using asterisk 1.0.9 on SuSE 9.3 (because I intend to grow the situation after this basic setup is functioning) The phone IPs are set to 192.168.0.201 and 202 respectively. I've had a look at iax.conf and extensions.conf but cannot see how to tie these IPs to an

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hi all, We use AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

Hi, Total newbie - I spent most of yesterday looking through various docs and could not come up with anything useful. I have 2 different SuSE 9.3 boxes - a P3 with 2 * ISDN BRI pci cards and a P4 with no asterisk relevant HW at all. I am using the default asterisk install that was loaded when I did a (full) fresh install of SuSE 9.3. I get the *same* error on *both* boxes, even though one of

Hello All, Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH? I keep getting 'Invalid Channel' but I cannot figure out why. Thanks! Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050607/b18b3743/attachment.htm

Does anyone know if FreePBX can be installed on a Linux box that was built using Asterisk@Home. I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only

Hello all, I am having trouble getting my IAX based Voip provider setup. Any pointers are welcome. So here is the deal. I am registered up and I can make outgoing calls but incoming calls fail. Configs all look good I thought. My PBX is behind our firewall with a direct NAT of one to one for an external IP. IAX port is forwarded UDP and TCP to the internal IP. * shows good registration and

Hi, I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the box (in a hope to sort out the uname -r issue mentioned below). I'm using the Asterisk Doc Proj vol 1 to guide me through the initial setup. I have no special HW and intend to use asterisk on an internal network just to get some experience. I have downloaded what I think I need and placed it in /usr/src (see

Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to

Howdy - I've been noticing a problem where I only receive a single call, before other calls go to voicemail. This only happens when the user is on the phone. I have the polycom 501's setup for 2 lines per key and 2 line keys for the first registration, which should allow for multiple calls. Anybody have any ideas what is going on? thanks Rolf Brusletto