search for: telephony

For those who are using SuSE: At last they've managed to create ready-to-run packages for openSUSE_11.1. They are there since a couple of hours... (For other versions it was allready available for some time on the OBS) /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/asterisk16-devel-1.6.0.6-82.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/libspandsp1-0.0.5pre4-59.2.x86_64.rpm /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/zaptel-kmp-default-1.4.12.1_2.6.27.19_3.2-32.2.x86_64.r...

Hello, I have ISA card LineJack. I could not find any information if this card can work as fxo with Asterisk. If it can work, can somebody point me how to install it on my Asterisk box. Or maybe there is some documentation about it how to install LineJack. I will be very thankful for any help. Regards Bartosz Jozwiak ------------------------------------------------- This mail sent through IMP:

Hi, I just came across this annoncement, which is particularly interesting as it is only 25 min away from my place... :-) Anyway, I guess the core of this is targeted at developers mainly. Cheers, Philipp Free Software/Open Source-Telephony-Summit 2004 http://www.guug.de/veranstaltungen/telephony-summit-2004/ http://www.heise.de/newsticker/data/avr-16.12.03-000/ We are happy to announce that the first summit on Free Software/Open Source-telephony solutions is going to take place from January 16th till 20th in Geilenkirchen, German...

Free Software/Open Source Telephony-Summit 2006 Tuesday, May 2nd 2006 Wiesbaden, Germany For the third time the German Unix User Group (GUUG - www.guug.de) organizes the Free Software/Open Source Telephony-Summit, an international workshop and technical conference for developers and u...

Hi, I'd really appreciate it if you can explain this to me. I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. I'm new to IP Telephony and telephony and general and I researched a lot but still confused about what I really need. I know that I can setup an IP-Telephony for my LAN using a SIP server and SIP compatible software phones. But the challenge is how can I connect to the PSTN so that I can send and receive calls? I came t...

For those of that are fans of stackoverflow.com, and stackexchange.com, there's an effort to define a telephony stackexchange site. It's still in the definition phase. What it needs to move forwards is more votes on on/off topic questions, and perhaps some better questions to vote for or against. If you're interested in helping out, or following the progress, visit: http://area51.stackexchange.co...

...(Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) __________________________________________________ So I made sure that the fxo is loaded first. Now it complains on channel 2. I also ran ztcfg -vv but no diff. So I ran dmesg which shows Zapata Telephony Interface Registered on major 196 No ISA tormenta card found at d0000 Zapata Telephony Interface Unloaded a few times. And: Zapata Telephony Interface Registered on major 196 PCI: Assigned IRQ 10 for device 00:06.0 Found a Wildcard FXO: Wildcard X101P PCI: Assigned IRQ 5 for device 00:09.0 also...

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression fix as described in issue #16268. Asterisk 1.6.0.19, and 1.6.1.11 contain an additional SDP regression fix a...

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression fix as described in issue #16268. Asterisk 1.6.0.19, and 1.6.1.11 contain an additional SDP regression fix a...

...l curl doxygen libnewt-dev mysql-client php5 php5-cli libmysqlclient15-dev libncurses5 libncurses5-dev openssl libssl-dev libssl0.9.8 mpg123 make g++ subversion subversion-tools newt-tcl linux-headers-`uname -r` php5-memcache php-pear DAHDI #cd /usr/src #wget <http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linu x-complete-2.2.0.2+2.2.0.tar.gz> http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux -complete-2.2.0.2+2.2.0.tar.gz [ <http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linu x-complete-2.2.0.2+2.2.0.tar.gz> ^] #t...

The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The related security advisory AST-2009-004 has been released along with this announcement. Please read that advisory for more information. The release candidates and betas, in addition to other fixe...

** Want to see who you're talking to? Video telephony is growing. A couple of developers has formed the Asterisk Video Task Force in order to improve the support for video telephony in Asterisk for the 1.6 release this fall. There is already support for video in the SIP and the IAX2 channel, but we need to add more in order to improve the support...

Hi, Is it me or unlike http://downloads.asterisk.org/pub/telephony/asterisk/, http://downloads.asterisk.org/pub/telephony/dahdi-linux/ is not showing lastest changelog anymore ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140714/80cefb92/attachment.html...

...burke.ca/asterisk-tfot-uncropped.pdf Sorry about the excessive noise, but I figured I should mention this. >Date: Mon, 20 Feb 2006 18:55:50 -0500 >To: asterisk-users@lists.digium.com >From: Alexander Burke <asterisk@alexburke.ca> >Subject: Download "Asterisk: The Future Of Telephony" > >Hello, list! > >I'm hosting a mirror of the book "Asterisk: The Future Of Telephony" >by O'Reilly Press, published under the Creative Commons license; I >believe this license allows me to do this, but if I'm mistaken, >please let me know. >...

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html http://downloads.asterisk.org/pub/...

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html http://downloads.asterisk.org/pub/...

...leases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, and 11.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve the following issues: * A remotely exploitable crash vulnerability exists in the SIP channel driver if an ACK with SDP is received after the channel has been terminated. The handling code incorrectly assumes that the channel will always...

...Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerabl...

...Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerabl...

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian insmo...