similar to: 1.4.22 vs 1.4.21.2 - IAX2 regression ?

Experts, Sorry if this is a FAQ but I''m not on this alias. Please reply directly to me. I''m working on a project setting up a web portal that will use 2 hosts for load balancing ftp''s. I wanted to use ZFS to showcase it to our customer. What I''ve been trying to setup is anonymous ftp to a host that is sharing a ZFS file system. Anonymous ftp is configured and

Hi I'm a new rsync user. I have two systems, A and B both with rsync installed. I've setup rsync in daemon mode on system B and want to rsync a directory /projects/sw/lib/ from system A to system B. My rsyncd.conf file on system B looks like this: [swlibrary] path = /projects/sw comment = Master Libraries If I execute on system A: rsync systemB:: I can see the

Hi, I want to automate some of the maclist and rule functionality: User connects to the network and gets a DHCP address from the shorewall box. Using squid and redirection, all the user can do is go to a login page on the firewall User logs in correctly to the form on the webpage and a process captures MAC and IP address info from the dhcpd.leases file Once authenticated, a maclist entry and an

Hello, We have a situation where a call comes in, users are notified via an external process (curl request to web service), and we can't answer the call until a callee can call in and pickup the call. How can we implement this functionality? We tried using : [caller-inbound-leg] ; code to send the CALL_UUID information to users. exten =>

Before I post to networking-discuss I wanted to ask if anyone had tried this on this: I''m trying to run a dhcp-server on a dom0 over a vnic so that the domU''s can get IP addresses. I created a vnic r1 over e1000g0 and gave it a static IP 172.0.94.111/24 so I can run the dhcp server over this vnic. root@lm2-dom0:~# dhtadm -P Name Type Value

I've been setting up a couple of * boxes with IAX trunking between them. But I've been seeing some strange IAX behavior. Asterisk version is latest CVS-04/21/04-18:10:19. Here's what I'm doing: the boxes are peers, and I have setup my iax.conf file to look something like this: << machine1 >> [iaxuser] type=friend username=iaxuser secret=foo auth=md5

Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use

Hello, I would like to know whether there is a way to associate a TV media server with Asterisk. Is it possible to access TV Chanels in the Telephone Sets. Anybody have any tips or documents related to this please let me know. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

After battling for days trying to figure out what was wrong with my iax.conf it was determined that I do not have any inkeys set on the digium server. Now whether that is something new or just in a few cases I am not sure. Messing around and reading on IRC and the mailing list I could get certain things to work and then break other things. Now I can dial a IAXTEL number, 800 number and FWD

Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, <unfinished ...> +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached On a second sister-machine with a mirror install we have the same problem. So it doesn't seem to be a

Hi, Ii try to connect an Asterisk server running 1.4.21.2 version with gtalk2voip services. Everything is fine till the call for DTMF test: there is no audio and Asterisk shows [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission SIPCALL-435578583-1984100284 at 72.20.112.114 for seqno 1 (Critical Response) [Nov 18 14:51:47] WARNING[20502]:

Hi guys.. I just wanted to know if this config could work correctly, since a lot of u guys have been working a lot with asterisk... Asterisk 1.4.21.2 -> chosen because is the last with Zaptel support ? Zaptel 1.4.12 Add ons 1.4.9 Currently i have a 1.6.1.0 Server running with no problems, but I'm about to buy a openvox a400p Card, so i would like to test both environments, 1.4 and 1.6,

The Asterisk.org development team has released Asterisk versions 1.4.21.2 and 1.2.30. Both of these releases include fixes for two security issues. Both of these issues affect users of the IAX2 channel driver. For more details on these vulnerabilities, see the published security advisories, AST-2008-010 and AST-2008-011. AST-2008-010: Asterisk IAX 'POKE' resource exhaustion -

The Asterisk.org development team has released Asterisk versions 1.4.21.2 and 1.2.30. Both of these releases include fixes for two security issues. Both of these issues affect users of the IAX2 channel driver. For more details on these vulnerabilities, see the published security advisories, AST-2008-010 and AST-2008-011. AST-2008-010: Asterisk IAX 'POKE' resource exhaustion -

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

Hello I'm using the Read() function to play a message prompting for the user to type a number followed by the # key to validate, with a 30s time-out and 2 tries: ============== [test] exten => s,1,Wait(2) exten => s,n,Answer ;typed DTMF: prompt for number to dial: 2 tries, 30s time-tout exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,,,2,30) exten =>

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o