similar to: US source for compatible ISDN cards?

I have made my first run at building FC1 and FC2 rpms for Asterisk 1.0.0. These are based off of Greg's spec files which a few minor changes. Source and binary RPMs are available at: ftp://asterisk.purplehat.net/pub/asterisk/ Please report any problems with these files to myself and not Greg as he has nothing to do with these builds. :) Sorry these took so long; it was a wild week at

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network

Hi, I have installed Asterisk 1.4 with mISDN with the install-asterisk.tar.gz script from beronet.com. On my system I have two cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to work well with mISDN on my system from a previous installation. Now however, the AVM card works well at first glance, i.e. it "registers" incoming calls and works through the asterisk

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

Can anyone work out why my callerid doesn't work on my isdn4Linux with asterisk (or without asterisk for that matter)... This used to work fine, and I am quite confident that the telco is sending callerid information (because they always do on all ISDN lines standard, only extra cost on POTS lines). This is the information from dmesg, whether asterisk is running or not: isdn_net: Incoming

Hi, I'm curious what chip Digium is using in the latest TDM400 cards. Specifically, to my recollection, they used to use the TigerJet 320G, however somewhat recently, Tigerjet was bought out, and now the 320G is no longer produced. Maybe a better question is: is there a way I can take the latest DAHDI source and get a list of supported chipsets from it? Thanks. MCH -------------- next part

I'm trying to get a basic Asterisk configuration together for ISDN incoming / outgoing calls. I have two Cisco 7905g phones working (at least talking to each other) and have purchased a NETjet-S PCI ISDN card for routing calls to / from ISDN. The state I've managed to get it to is:- -- Executing Ringing("SIP/PHONE2-d557", "") in new stack -- Executing

Hi Everyone. I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer >From the sdp can anyone suggest why secure audio cannot be provided ????v=0 ????o=- 6611325078116277019 2 IN IP4 127.0.0.1 ????s=- ????t=0 0 ????a=group:BUNDLE audio ????a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l ????m=audio

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

Hello everyone, I have been using an AVM Fritz! card with chan_capi and mISDN for quite a while in PTM mode and it was working finely. Now, I needed more DID/MSN, so I switched to PTP. But now nothing works anymore :( I am using Asterisk on Debian Sarge stable and installed Asterisk along with chan_capi from apt-get. I installed mISDN from the CVS of isdn4linux.de. It is : - Asterisk

What would you suggest for a usb phone? I'm thinking a real phone - not an adapter. The kind of thing which is small enough so you can take it on a trip to place VoIP calls from your laptop. Something that ties into asterisk on linux and (more importantly to be honest) a softphone on XP or even its own client. Browsing the web I found that most places either sell them by the truckload or

Hi all, I am trying to add a TigerJet usb adapter to my asterisk installation I have the 1.2.4 asterisk bristuffed version, holding two zaphfc (ISDN) cards If I connect the TigerJet adapter to my linux box (Suse 10) i see: Bus 005 Device 001: ID 0000:0000 Bus 004 Device 003: ID 06e6:831c Tiger Jet Network, Inc. Bus 004 Device 001: ID 0000:0000 Bus 003 Device 001: ID 0000:0000 Bus 002 Device

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I installed my TDM400P into the PC, it's really slow to boot now, when it finally does I gets stuck in a loop of reporting "isac xdu no tx_busy". Anyone able to assist? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

What's the impact on encoded speech quality (per given bitrate) when the encoder cpu complexity is reduced all the way down for Opus? Rather, how big is the impact? Secondly, can someone comment on wideband speech quality comparison between Opus and iSAC with and without the cpu complexity of Opus turned all the way down? Thanks! -------------- next part -------------- An HTML attachment was

Hello again, the same can be done with textpad and adjustments in preferences/file also changes owner when a file is changed and saved. greetings jens Kramer -----Urspr?ngliche Nachricht----- Von: Willem Jaap Zwart [mailto:W.J.Zwart@NescioLudens.nl] Gesendet: Freitag, 8. April 2005 16:55 An: Kramer Jens ZFF ISAC Cc: samba@samba.org Betreff: Re: [Samba] Samba 3.10 and higher Hi We noticed

> I am asking this because it is believed that Skype is using some iLBC and > iSAC since GlobalIPSound listed Skype as a partner. I think (from what I've heard) that's what Skype uses. I have no idea how iSac sounds because it's proprietary and I've never used Skype. Jean-Marc > > Thanks, > Joe > > -----Original Message----- > From: Jean-Marc Valin