similar to: Found unknown media description format

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]:

hi, it turns out that the iksemel library (which i installed using an rpm) was returning 0 when the function iks_has_tls() was called. it should return 1 otherwise res_jabber.o thinks gnuTLS is not installed. i confirmed this by running a test program i wrote, that calls iks_has_tls . it returned 0. i downloaded iksemel source, compiled it and now the test program returned 1. now,

I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gtalk. The Phone rings and connects - but no audio! I am using a self-compiled asterisk 1.4.2 There is a lot of output on the CLI but I can't make sense of it. Perhaps somebody can help? Michael Output from the CLI: JABBER: gtalk_account OUTGOING: <iq

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

Hello, I need to do an intersection between the list elements (partitionslist) and the columns and rows of a matrix (mm), so that the result will be the sums of the rows and columns. Thanks a lot, Romildo Martins Example 1.The Intersection and sum betweeen partitionslist[[1]][[2]] and mm is indicated in bold. 2.The Intersection and sum betweeen partitionslist[[1]][[2]] and mm is indicated in

> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want >

I am trying to use: ocfs-support-1.0.10-1 ocfs-2.4.21-EL-smp-1.0.11-1 ocfs-tools-1.0.10-1 with RedHat AS 3.0, 2-node cluster with shared SCSI. 2 dell 1650s, dual CPUs, PERC 3/DC cards chained to a PowerVault 220S. I am using lvm, and here is my layout: [root@cluster1 archive]# df -h Filesystem Size Used Avail Use% Mounted on /dev/sda2 32G 5.1G 25G

we started using an ocfs file system about 4 months ago as the shared archived redo destination for the 4-node rac instances (HP dl380, msa1000, RH AS 2.1) . last night we are seeing some weird behavior, and my guess is the inode directory in the file system is getting corrupted. I've always had a bad feeling about OCFS not being very robust at handling constant file creation and deletion

Dear all, First off, I am very new to asterisk so forgive me if any of my comments or questions seem trivial. Thanks to [this post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/) and [this post](http://www.davidvossel.com/?p=28), I have GV set up on asterisk through jabber.conf and gtalk.conf. I can successfully dial out from asterisk. I'm trying to set up an

Am trying to do restore of control file on Linux Machine 64Bit CenTOS5.2, i get the following RMAN> restore controlfile; Starting restore at 29-07-2008 using channel ORA_DISK_1 channel ORA_DISK_1: starting datafile backupset restore channel ORA_DISK_1: restoring control file channel ORA_DISK_1: reading from backup piece /u01/oracle/product/10.2.0/db/dbs/c-2142365377-20080729-01 ORA-19870:

I've put a t100p in our * server and I'm having trouble configuring it. It is directly connected to an Adtran TA 750 channel bank with two FXO cards (8 analog incoming lines total). I'm able to insmod and modprobe both zaptel and wct1xxp with no trouble, but when I start * with /usb/sbin/asterisk -c I get the following output: [root@rosella root]# /usr/sbin/asterisk -c Asterisk

I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com

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Hello, I run a samba (v3.0.10) PDC and openLdap server on the same machine. They work fine, except sometimes the smb server cannot contact the ldap server. From a windows client, it looks like the server gets stuck, and it takes a few minutes before the shares are available again. In the /var/log/messages file, i get these error messages :

The Wine development release 1.5.25 is now available. What's new in this release (see below for details): - Proper cursor support in the Mac driver. - Fixes for right-to-left support in RichEdit. - Initial version of a Wingdings font. - Various bug fixes. The source is available from the following locations: http://prdownloads.sourceforge.net/wine/wine-1.5.25.tar.bz2

I was able to follow the search recipe in Chad Fowler''s upcoming book but I''m wondering if there is any sort of recipe/guide out there to actually perform the search once the auto-complete does its stuff. Anyone? It''s kind of hard to search for search.. -- Posted via http://www.ruby-forum.com/.

qemu in SLIRP mode offers DHCP services to the appliance. We don't use them, but use a fixed IP address intead. This changes the appliance to get its IP address using DHCP. Note: This is only used when the network is enabled. dhclient is somewhat slower, but the penalty (a few seconds) is only paid for network users. We could consider using the faster systemd dhcp client instead. ---