similar to: Issue with Hamlet ISDN PCI card(Cologne Chipset)

I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061028/87014bd9/attachment.htm

Hello, I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands) >From the moment i switched all inbound calls are terminated after aproximatly 1 minute. The provider tells me it's not their issue since I have no other configuration than all their other users. What can I do. I removed all asterisk functionality by forwarding the inboud call directly to a local

Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card.... We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then

Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not

Hi, I have an old partport optical scanner (an LG Scanwork 36a)... Of course,I can't use this scanner with linux (no driver). So, I'm trying to install win-driver with wine. :-) I read documentation (unfortunately insufficient) and... I fell in an "Hamlet' question"! The wine user manual asserts that I can install scanner's win-driver in wine, unfortunately with too

Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec???? Office A --------T1 ----- Astrisk TE05P----------------IAX2----------------Astrisk Box -2 | |

>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Message: 16 Date: Wed, 18 Oct 2006 16:10:38 +0100 From: "Neil Tancock" <neil@safeharbourit.co.uk> Subject: [asterisk-users] IAX Terminal To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

I have successfully placed a call into and out of the card from Asterisk using vISDN. The current vISDN snapshot now contains the PCI id's for the card so no patching should be required. Most of the initial testing failure was due to misconfiguration of vISDN by me and a bad entry in my sip.conf resulting in one way audio. So I have established that at least 1 channel of 1 port works in TE

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hi everybody I'm trying to make Asterisk 1.2.1 run under VMWARE and Suse 9.2. I use ZTDUMMY module for timing and ZTTEST gets an average precision of 98,4 %. Is there any way to improve it? Best regards Mauro Zanin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060117/945d544a/attachment.htm

Hi dear group members, I have finally an Asterisk box working, capable of receiving and making calls. I have this issue while calling mobiles from our SIP softphones: -------------------------- linux*CLI> -- Executing NoOp("SIP/2000-6850", "3487024125") in new stack -- Executing Dial("SIP/2000-6850", "ZAP/g1/3487024125") in new stack -- Called

Hi, I have a strange problem on some isdn modem channels. (* 1.0.9 / chan_modem / 2xHFC-S cards). Everything works fine but when the 2nd (and 3rd etc..) call comes in and * answers and there is about a 1/2 second of sound from the previous call (ivr) before the sound from the new call is heard. It just sounds bad and is quite annoying. I am assuming this is sound that is still in a buffer

Hi all I don't manage to get asterisk 1.2.5 or 1.2.6 running with the zaphfc driver.... The scripts from junghanns.net do download a very old libpri and asterisk version which is too buggy for me to use. Isn't there an acutal patch to get zaphfc support in *? -Benoit-

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

I may have found a source of an A-Ticked HFC 4BRI PCI adapter in Australia, and will be testing one next week if all goes well. I don't want to post the details of the reseller online unless invited to do so, so if nobody replies and says they are interested then I won't :) I'll follow up once I've tested it. Let me know if you want the details. James

Questions are about <1b> NetBios type record name for workgroup (for workgroup, not for node) . There is local subnet with Samba4 as DC (either Win NT or Win NT+) for mdom.net domain, and as following, there is mdom workgroup in neighborhood. But there are some other domain and/or non-domain workgroups in the subnet which Samba4 does not belong to. Is <1b> had to be registered by

Hi everybody, I managed to install card over Suse 9.2, I substituted Zaptel drivers and compiled them. Now "ztcfg" says I have one card with correctly configured 31 channels, but red led on back of card doesn't flash. Suse 9.2 has detected the card as a Tiger Jet card, since the chip on it is a Tiger 320. The second card configuration is still waiting for configuration, but I think

Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg "Im-sorry&an-error-has-occured" and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some