similar to: Asterisk 1.2.7 Page()

I am trying to page my Grandstream GXP-2000 phones and keep getting the error message: Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete destination '' supplied. How can I fix this error? The two contexts below do either one-way paging or two-way paging to all Grandstream phones in a list. [One_Way_Page_GROUP] ; one to many page exten =>

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC3581 sip users (type=friend) are in sip.conf. I have nat=no

Hi everyone: I'm trying to compile Asterisk on FreeBSD 6.0-RELEASE and I'm getting the following error: cc -O2 -fno-strict-aliasing -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -DMAKE_VALGRIND_HAPPY -I/usr/local/include -L/usr/local/lib -I/usr/local/include/spandsp -DZAPTEL_OPTIMIZATIONS

I'm trying to use a simple page function. It starts a MeetMe conference with the devices I've listed, but the devices hang up after 3-5 seconds. After doing some research I found this was a problem, and I needed to remove a (5) from app_page.c Well, my app_page.c didn't have the (5). I did make clean; make install again just in case I had some weird compiled version installed that

using app_page on asterisk 1.6.1.6, as documented, the 'q' option only determines if the caller is sent a 'beep' tone when conferencing. is there a way (existing or someone sending me a patch) to also make app_page beep all of the extensions being called? someone adding an 'a' (announce tone) parameter to app_page would be perfect. with auto-answer turned on with my

I am instaling asterisk on Fedora core 3. I have instaled zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error: .................... _GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_zapscan.o app_zapscan.c gcc -shared -Xlinker -x -o app_zapscan.so app_zapscan.o gcc -pipe -Wall

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. Here's the REFER that the phone at 2944093 sends directly to Asterisk: U 216.186.128.68:5060 -> 216.186.142.203:5060 REFER sip:3254102@216.186.142.203 SIP/2.0. Via: SIP/2.0/UDP 216.186.128.68;branch=z9hG4bKba3b074892377BD1. From:

I just realised my problem seems to be related to bug 0003710 - "0003710: [patch] Consultative transfers between asterisk servers". It's unclear from the bug info if this problem has been resolved yet. Anyone know? Doug. > -----Original Message----- > From: Douglas Garstang > Sent: Monday, March 27, 2006 4:41 PM > To: 'Asterisk Users Mailing List - Non-Commercial

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. It appears this related to bug 3710. It's unclear from the bug if the problem has been fixed or not. If it hasn't, then this seems pretty serious and would I guess affect any NAT-ted phones ability to transfer calls. Here's the REFER that the phone

I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely.... someone else must have seen this? hermes*CLI> sip show channels Peer User/ANR Call ID

Hi All, I have the latest AsteriskNow installed (1.5) and after a couple of months with system in production I have a need to use the Paging/Intercom features. I have the module installed and I am able to successfully intercom with individual phones using *80xxx (extension number) but if I create a paging group it does not work. I receive a message that it is an invalid conference number. I

I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently? exten => 5555,1,NoOp(${CALLERID}) hestia*CLI> -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack == Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN' This

I had an incident, whereby the caller didn't either hang-up their SIP phone properly or the disconnect/hang-up information didn't properly find their way back to the Asterisk-PBX and it left the company phone system in intercom mode with about 90 phones overnight (624mins, CPU utilisation was running much higher than normal until i used the meetme kick <channel> all

Hi, We just upgraded from 1.2.7 to 1.2.12.1. Everything is fine, except that asterisk seems to just crash at random. Often I can make it crash by using the ChanSpy function (which we use to monitor agents). Sometimes it will just crash on its own. The reason we were initially running 1.2.7 was because of the stability it gave us (weeks without a restart). We upgraded to 1.2.12.1 because it

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

The Asterisk.org development team is proud to announce the release of Asterisk 1.4.23. This release is available for download from http://downloads.digium.com/. This release is a significant bug fix update for the 1.4 release series. The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help! * Fixed

Hi, I was trying to follow this guide (http://libvirt.org/compiling.html) to compile libvirt1.2.7 on CentOS6.4. There is an error on executing "make". Below is the steps to reproduce and the error messages. Any help is appreciated. Thanks. $ sudo yum install gcc make gnutls-devel device-mapper-devel \ python-devel libnl-devel yajl-devel \ libxml2-devel libpciaccess-devel $ wget

I'm getting this Panic with some users on dovecot-1.2.7: Panic: file maildir-uidlist.c: line 1242 (maildir_uidlist_records_drop_expunges): assertion failed: (recs[i]- >uid < rec-> uid) There's another dovecot-1.2.3 running on identical hardware accessing the same NFS mail storage without problems.

I'm trying to build hivex 1.2.7 on Ubuntu 10.04, and get the following failure: Making all in xml make[2]: Entering directory `/opt/sandbox/src/hivex/hivex-1.2.7/xml' CC hivexml-hivexml.o CCLD hivexml hivexml-hivexml.o: In function `main': /opt/sandbox/src/hivex/hivex-1.2.7/xml/hivexml.c:86: undefined reference to `rpl_getopt'

Hello, I am currently working on Redhat 6 - Kernel 2.6.32-358.el6.x86_64. It has by default libvirt 0.10.2. I wanted version 1.2.7. So I downloaded this and libvirt-1.2.7.tar.gz and I followed the following steps: 1. ./configure (with its default settings) 2. make 3. make install Currently, the installation is in: /usr/local and the source tree is in /export/home/libvirt. I stopped the