similar to: mISDN (HFC-S) and TDM400P - isac xdu no tx_busy

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

All, I have not found/seen a resolution to the issue where my TDM400P seems to cause problems, as outlined in the "mISDN (HFC-S) and TDM400P - mISDN: ISAC XDU no TX_BUSY" thread. I have also not found/seen a simple 'how to' on patching DAHDi with ZAPHFC as outlined in the "HFC-S card" thread. Do I have anyother options with ISDN? Thanks in advance!

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

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

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

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

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

Hi All, I'm trying to use a HFC chip ISDN modem with mISDN and chan_misdn. The card is configured to NT, PmP mode. One Siemens ISDN phone connected to the modem. When I call the ISDN phone is called everything is just fine, but when I call from the ISDN phone I face some problems. - There is no dial tone when I pick up the handset. - I call a number but there is no ringtone and after I

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

Does any one put a HFC-S card working in nt ptp mode? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100221/ca845019/attachment.htm

Thomas, Thank you for your interest in this, you description is as accurate as I can see. > From my perspective your challenges will be to get the containers right. > WebM for audio+video > Ogg for audio > > Also (I'm not that familiar with webRTC) you might need to reencode > to Opus and VP8 in some cases? here is the great news

Hi! everybody: We used opus-codec for a VOIP gateway. The GW is running at a UBUNTU server. The opus stream is transcoded to G711 pcmu stream.So there are many opus codecs running simultaneously. We noticed that if there more than 5 streams in. the voice then has notisable glitchs.More streams in, worse voice got. Then we write test code for opus-codec which encode a .pcm file simultaneously.

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

Hello, I am having problems, getting my asterisk-domU to work properly. It consists of the following components: - Debian Etch under Xen-3.1 with a 2.6.18-kernel - Asterisk 1.2.24 - mISDN-1.1.5 I have 2 HFC-ISDN-cards, which I pass through to the Asterisk-DomU in permissive mode. This is working fine. The strange problem is, that the two HFC-ISDN-cards are not beeing initialized by the

Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]:

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

Hey all, So we have been advised from this thread https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 to not use http put as it is not in real-time, instead they are suggesting the use of SDP, is that something that icecast supports? Or does anyone have other ideas on this? ~stephen On Sun 12 May 2013 01:51:31 AM CDT, Thomas Ruecker wrote: > Hi, > > On 11

Due the software Ultraedit, it is possible to manipulate the ownership of Files!!! This may be a big securetyhole. A test.txt owner jens:group fish Unixrights 760 opened an manipulated whit ultraedit, saved. ther will be 2 Files one test.txt which is owned by the modifier( e.g hans:fish), and a test.txt.bak which is owened by jens.fish. That's OK. so if i repeat this sequenz again, i will

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read