similar to: mISDN & Asterisk 1.4: HFC-S card not responsive

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 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

Full_Name: Manikandan Narayanan Version: 2.8.1 OS: linux-gnu Submission from: (NULL) (155.91.28.231) Hi R-folks, I have two three-line text files: tst1, tst2 (they are the same except that the second line is longer in tst1; see cat() cmds below). read.table is only able to read the 3rd line in tst1, however reads tst2 correctly as shown below. This happens both in R 2.5.1 (windows) and R

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

ioemu: fix xenfb slow case update Signed-off-by: Samuel Thibault <samuel.thibault@eu.citrix.com> diff -r 067d8f19e78a tools/ioemu/hw/xenfb.c --- a/tools/ioemu/hw/xenfb.c Thu Feb 28 13:55:37 2008 +0000 +++ b/tools/ioemu/hw/xenfb.c Fri Feb 29 15:25:17 2008 +0000 @@ -1072,7 +1072,7 @@ /* A convenient function for munging pixels between different depths */ #define

Hi, i use an BN4S0 with misdn an asterisk on Linux 2.6.9. The hfcmulti module is loaded with option: type=0x04 protocol=0x2,0x2,0x22,0x2 layermask=0xf,0xf,0xf,0xf and the fourth port is connected to an ISDN PTMP (MSN) port. Call to #72 from S0 (BN port 4) are not accepted from asterisk but why ? Can anyone give me a hint ?? misdn debug messages follows: lib: NEW_CR Ind with l3id:80001

Hi I have beronet BN8S0 isdn card in my asterisk and , card is working fine, but when I try to dial to special number 118913 ( telephone number information) from polish telecom TPSA, I always geting timeout . Bellow is isdn signaling dump : --> * CallGrp: PickupGrp: --> rxgain:0 txgain:0 --> * dad:118913 tech:mISDN/2-u25 ctx:default --> * Setting Context to from-tpnet -->

Hi. I'm running Asterisk server with 10 sip phones, and 2 grouped T0 lines with 10 DDI numbers. My provider is France Telecom and my setup is : - Debian Lenny - Asterisk 1.4 - Linux kernel 2.6.25.17 - mISDN 1.1.8 driver - Sip phones Thomson ST2030 No problem with the SIP . But when reveiving a call on RNIS line (any of the DDI numbers), the associated SIP phone rings indicating _two_

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,

Greetings I'm running Red Hat 6.2. Both smbd and nmbd are up and running. I can see my Linux box from PCs in my LAN. Problem is, I can't access them. I tried the troubleshooting guide found at http://us2.samba.org/samba/docs/DIAGNOSIS.html, and got as far as the second step. When I do "smbclient -L myserver," I get the following error message: session request to ODYSSEUS

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

Not a problem ... sharing a solution (this time)! Please correct my understanding of the process, if required. "i_stream_read() failed: Permission denied" is an error message generated when a large-ish file (>128kb in my case) is attached to a message that has been passed to Dovecot's deliver program when SELinux is being enforced. In my case, these messages are first run

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

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

Hi One of my softphones is supposed to support g711 , however I am getting these errors and a 404 not found when I try to make a call from it. However on xlite it works fine using g711. Below is the log of the phone that is not working. Content-Type: application/sdp Content-Length: 1123 P-hint: outbound v=0 o=- 1218448446 197568495 IN IP4 127.0.0.1 s=- c=IN IP4 192.168.0.176 t=0 0

facing problem with originating webrtc calls 1-when iam doing call from webrtc iget ice working <--- SIP read from WS:91.196.158.205:1466 ---> INVITE sip:0669197533 at 77.91.132.9 SIP/2.0 Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315 Max-Forwards: 69 To: <sip:0669197533 at 77.91.132.9> From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43 Call-ID: