similar to: SIP Changes???

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

Hi Everybody, Has anybody tried the above (or indeed any other 4XBRI cards) successfully with Asterisk. As far as I can see the above mentioned card is an active ISDN card but supported by it's own I4L driver. This leads to interesting questions particularly regarding echo cancellations (which usually doesn't work on the cheap passive cards with one exception as far as I can see).

Dear Sirs, I've got a weird problem with IAX2 transfers. My setup consist of 3 Asterisk servers. One is located in Europe on a public IP and a local PSTN connection through ISDN. Two are located in South-east Asia - both on public, but dynamic IP. These two each have a bunch of SIP phones attached. All 3 systems are running latest CVS (ok - might be a day or two old) on Linux 2.6.6 with

Hi Chris, I know this is quite an old email, but I was browsing through the archive :) I am currently working on "embedding" asterisk in one of Allwell's STB's. The idea is more or less exactly like yours. The STB will be solid-state and contain OS, Asterisk, Basic configuration and voice files on a flash disk. It will boot up and get a network share via DHCP. This network

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN

Hi! I have some problems with the download of Asterisk-addons. I try to follow instructions that I found in http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql , but nothing to do. This is my shell: [root@obelix root]# cd /usr/src [root@obelix src]# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot [root@obelix src]# cvs login Logging in to

Hi Everybody, I've been asked to upgrade an old NT4 server with a new server running Debian/Samba. I've got no problem with migrating the old server, but I do have one "unknown". The company mail run on an Exchange server that is most likely part of the NT4 network. Has anybody tried this setup? I did look in documentation and google and found precious little - which to me

Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

Hello, I just noticed an error in the current version of Klaus-Peter Junghanns bristuff package, especially in the HFC module. Everytime I try to unload the HFC module with "modprobe -r" I got a kernel panic and the complete server hangs up so I need to do a hard reset. Regards, Julian Pawlowski

I'm having issues with compiling libvirt on OSX 10.12  ~/Development/github/libvirt/ [master] make /Library/Developer/CommandLineTools/usr/bin/make all-recursive Making all in . Making all in gnulib/lib /Library/Developer/CommandLineTools/usr/bin/make all-am make[3]: Nothing to be done for `all-am'. Making all in include/libvirt make[2]: Nothing to be done for `all'. Making all in

We are looking soon at buying a system to deploy asterisk as our company's PBX. We run SuSE here and like it and our asterisk test platform is SuSE 9.0 with the 2.4 kernel. Is anyone running * and the zaptel drivers under SuSE 9.1? With the 2.6 kernel? Is * 64-bit safe (i.e. no 32bit assumptions in the code) so I can run it on an AMD Opteron in 64-bit mode (with whichever kernel is

Hi all, I've run into an issue we have with dovecot versions 2.0.12 through 2.0.14 on latest CentOS 6.2 when dealing with Maildirs located on NFS partitions. Although I saw Mr. Egbert's thread, his resolution does not work on our end. When dealing with users with over 500 messages in any of their Maildir folders, they occasionally receive an error from the server: Internal error

> I'd like to create the following: a user picks up the phone > (gets a dial tone), dials '0' for an 'outside' line, gets a > second (different?) dialtone, and is able to enter an > external phone number. Klaus-Peter Junghanns has something like this on his page: http://83.137.99.170/jn/relaunch/asterisk/page19.html It didn't work for me correctly so I

Hi all ! We have 3 NTBAs which are all going to our existing PBX. Our areacode is 06003 and our DDI enabled number 9141. I want to exchange that PBX with Asterisk, but still struggle to get it working. My CAPI.CONF is currently like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.800000 txgain=0.800000 [interfaces] context=in-capi msn=60039141 incomingmsn=* controller=1

http://www.blorp.com/~peter/in_vorbis_test.zip report any bugs -Peter --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the list will be

I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more

http://www.blorp.com/~peter/zips/in_vorbis.zip another bunch of bugfixes and some new features (see tag editor advanced mode) -Peter -- 26 maja - Dzien Matki. Wygraj kwiaty dla swojej Mamy! Do rozdania 75 bukietow z dostawa i zyczeniami. [ http://zakupy.onet.pl/prezenty.asp?k=7 ] --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

Hi - Anyone using zaphfc cards (bri-stuff-0.0.2) on a Portugal Telecom BRI service? I am getting loads of errors and the audio is very distorted all the time. System is a Dell PIII/933 MHZ with SCSI. Here in the UK we have perfect audio with a Compaq PIII/500 MHZ SCSI system. Files are identical in both systems as below: Is there anything magic about Portuguese ISDN? The PCI bus in the