similar to: UK install

Chris you might know the answer to my HUUUUUUGE problem A few weeks ago you posted this message: "I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works)

Hi all, I have an installation where each user on subnet 192.168.1.0/24 is connected via a multiplexer. The problem is that if any client uses more than about 48Kb/s, the multiplexer crashes. I need to limit each client to under this rate, say 32Kb/s. I have seen examples on creating a class per host but is there a simple way of saying "any host from 192.168.1.0/24" so I dont have

Advance apologies for the length of this mail; I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works) and all my SIP extensions (which all work) along with

All, I'd like to buy an E100P Wildcard frm Digium, but i prefer to buy it in Europe (costs, ...). Could somebody point me to an european reseller ? Thanks in advance. -- Nicolas Cartron nc@ncartron.org

I have been experiencing hung up when answering incoming calls through x100p. NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered).. -- Executing Wait("Zap/1-1","1") in new stack -- Executing Answer("Zap/1-1","") in new stack -- Executing DigitTimeout("Zap/1-1"."5") in new stack -- Set digit timeout to 5 --

We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make "outbound" calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we

Hello. I am totally new to this PBX/asterisk/telephony world. I understand that what I need between the T100P/E100P and the POTS analog lines is a channel bank/multiplexer? I am looking for recommendations on this type of hardware. I have been looking around and found a product from valiant communications (http://www.valiantcom.com/) and/or

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18

I am getting to my wits end with these phones (and so is my boss). I am getting an random echo on these phones and I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made

Dear all, Is there an another way, rather than using metainfo.yaml, to add links to external websites and files to the navigation menu? I would prefer something that I could configure on a per directory basis. Best wishes, Marc Argent

Once in a while, someone mis-dials when transfering a call on their Snom SIP phone (using the Transfer button). Instead of sending them to, say, 1940; they mistype and enter 194 or 190 or somesuch. This ends up on the PSTN (for which three digit calls are valid); not what anyone wanted. On our old PBX (Network Alchemy Argent Office) there was a dialcode that grabbed back the last call that went

Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end

Hi All, Has anyone hooked up * as an extension/trunk of an Avaya system that has around 2 ISDN30e's. Trying to add 100 extensions to one of our systems, but not sure where to start reading. Thanks. -- Kind Regards, Gavin Henry.

Eeeeck! Asterisk is inserting random digits in dialed numbers. So far I've seen it insert a 2 after the STD (area) code and insert an extra 6 or 7 in the STD code. It's pretty repeatable although the inserted number changes. My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02). There's an ISDN PBX on the second span and a BRI euroisdn on the first. Calls from the

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

If you are really wanting to build something on Raspberry Pi or similar ARM platform, you could also take a look at Elastix for ARM. http://www.elastix.com/en/downloads/ Elastix is a fully integrated platform, and includes the majority of necessary components in one installation. The new Raspberry Pi 2 platform may be perfect for your needs in this respect, although based on your load, the B+

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940, but not when using a POTS phone plugged into a TDM400 FXO port. It doesn't seem to matter if the call

Hi All, I would like to set up a samba server but using the same user / pass for unix logins and smb logins. This works fine if I use non-encrypted passwords but I have to apply the registry patch to set my win98 clients to use non-encrypted passwords. Could I use pam_smb to authenticate the Linux box against its own smb server then use encrypted smb passwords? I understand that if the smb

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