similar to: asterisk doesn't know version of asterisk-addons?

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

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

Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, why udev's for board in "/dev/zap"(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : )

Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure

hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/86e4ad95/attachment.htm

Hi! I installed shorewall shorewall-2.1.7-1 on a newly installed box. When using 1.4 versions I could enable shorewall in ntsysv, now shorewall doesn''t show up in ntsysv? I am running shorewall on a RedHat Enterprise Linux rebuild (www.taolinux.org). Is this a bug or a feature?

Hi list! Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk? Any experiences / caveats? If anyone would be willing to share the dump of their IP600 config file, i would really appreciate it. Is there anything special i should put in my asterisk config? Thanks !!! Remco

Hi, Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this: Note: MySQL libraries are installed and the structure is as follows: /usr/src/astsources/asterisk-1.2.13 /usr/src/astsources/asterisk-addons-1.2.5 in /usr/src/astsources/asterisk-addons-1.2.5 I do: make clean make and the output is: ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`

Could anybody help me with the correct rules to allow a client to connect to an NFS share on my Linux box/firewall? >From what I understand from some howto''s on the net is that there are several daemons that are required to make NFS work. Some rules I found in the NFS howto: iptables -A INPUT -f -j ACCEPT -s 192.168.0.45 iptables -A INPUT -s 192.168.0.45 -d 0/0 32765:32768 -p 6 -j

Hi! I have this rule to allow incoming VNC connections to a host on my private LAN: DNAT net:$SITE1 loc:$JOHN tcp 5900:5910 This works perfectly the host can connect without problems. Would it be possible to link the internal host to an external ip? For example if Site1 would connect on the VNC port, all traffic gets forwarded to loc:$JOHN. But if net:$SITE2 connects

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

The information (or lack of it) on upgrading from zaptel to that @&*^QW%&^%!!! dahdi is very frustrating. I cannot find anything on how to uninstall zaptel, i found an earlier post to this list which suggested make uninstall and make remove in the zaptel directory which just generates errors and does nothing (on zaptel 12.1). Then i install dahdi-linux and dahdi-tools and i want to

Hi list! I've been bitten by a bug in the latest kernel 2.6.9-34.EL and I already removed the old kernel. Are old kernels retained at any place? I need to go back to 2.6.9-22.02 but where can I find it? In the Centos 4.3 directory I find only the latest kernel, not the older ones? Thanks!!

hello, I tried to compil asterisk-addons-1.6.2.0-rc1, and I have that error: [CC] res_config_mysql.c -> res_config_mysql.o res_config_mysql.c:1367: error: unknown field ?update2_func? specified in initializer res_config_mysql.c: In function ?parse_config?: res_config_mysql.c:1432: error: ?CONFIG_STATUS_FILEMISSING? undeclared (first use in this function) res_config_mysql.c:1432: error:

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

Did the following: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel svn checkout http://svn.digium.com/svn/libpri/trunk libpri Compiled and installed zaptel, libpri, asterisk and finally asterisk-addons. Following errors ocurrs when compiling

Hi all I don't know what went wrong but i no longer seem to be able to compile asterisk. I first do : cd /usr/src/dahdi-linux-2.1.0.4 make clean ; make all ; make install cd /usr/src/dahdi-tools-2.1.0.2 ./configure ; make clean ; make all ; make install ; make config So far.... so good but then when i do : cd /usr/src/asterisk-1.6.0.6 make clean ; ./configure ; make all ; make install i

Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone

Hi list! I'm totally fed up with bristuff (or it's instability with a simple HFC-S card), 2 out of 3 times when people try to call they get the information tone that the number is not connected. I would like to try vzaphfc and I am looking for information on it. >From previous posts I found that the only place where the sources seem to be maintained and available is at the debian

I have a CentOS release 4.7 box running asterisk-1.4.26.1 with dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0 I regularly get these messages, is this something i should be worried about? [Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Aug 21 01:05:07] ERROR[4343] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory [Aug