Displaying 20 results from an estimated 1000 matches similar to: "Extension Status"
2010 Feb 24
2
Problems in Asterisk Real Time (Urgent help )
Hello,
Asterisk Real time database worked on astersik 1.6.2.0 but now i am working
on Asterisk to latest version which is 1.6.2.2 ,there is a a warning
[Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping
for 'sippeers' found to engine 'mysql', but the engine is not available
[Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register:
2010 Jan 06
1
Fastagi-mapping problem
Hello
I am new in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started the
first example Hello AGI in this tutorial
http://asterisk-java.org/development/tutorial.html I put the jar file and
the proparty file in folder
i wrote in extensions.conf this exten => 1300,1,AGI(agi://
192.168.50.127/hello.agi,${EXTEN},${UNIQUEID},${CALLERID(name)})
I started AGI server ,then when i call
2010 Jan 27
1
Asterisk Database Configuration
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`ipaddr` varchar(20) NOT NULL default '',
`port`
2010 Jan 28
0
Database Configration
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`ipaddr` varchar(20) NOT NULL default '',
`port`
2010 Jan 28
0
Asterisk Database
Hello
I am trying to attach a database to asterisk , can anyone help me?
in extconfig.conf sipusers => mysql,general,sip
in res_mysql.conf [general]
dbhost = 192.168.50.125
dbname = asterisk
dbuser = root
dbpass = ahmed
dbport = 3306
dbsock = /tmp/mysql.sock
i created a table in MySql
CREATE TABLE `sip` (
`name` varchar(40) NOT NULL default '',`username` varchar(40) default
2004 Dec 06
1
iax2 nativ bridge question?
hallo all,
i would like to know, as i would suspect, nativ bridiging should work also,
if only one iax party is behind an nat router and the other has a public
ip. when i connect to iax clients, which have both pubic ip's nativ
bridging is working. if one of the clients is behind an nat, the iax2
channels always get routed through the asterisk server (latest stable
version from cvs) ?? i
2014 Dec 30
3
status - Unmonitored, how to change it
How to change status of peers "Unmonitored" to monitored?
Home users showing "Unmonitored" some display timing.
Name/Username Host Mask Port Status
zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored
clinic_server (null) (D) 255.255.255.255 0 Unmonitored
voip
2005 Mar 04
2
Broadvoice + incoming call works only for ~2 minutes
Hi, all.
The asterisk setup is working fine, receiving calls via broadvoice "initially". ?
When call comes in via broadvoice number, asterisk picks it up and routes
correctly, as long as the call came in within ~2 min from the previous one.
In other words, as long as a call comes in within ~2 min since the previous one,
asterisk will answer the call. However, if the call comes in
2007 Jul 08
2
Auto Fall Through when kicking users in MeetMe
Hi all,
My scenario is such that I have three users connected to a conference.
CLI> meetme list 1234
User #: 01 9176502096 <no name> Channel: Zap/23-1
(unmonitored)00:00:32
User #: 02 john john Channel: SIP/john-b7800468
(unmonitored) 00:00:28
User #: 03 6463875998 <no name> Channel: Zap/22-1
(unmonitored)00:00:19
3 users in that
2015 May 28
4
Peer is UNREACHABLE
Hi list!
I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a proxy to
other SIP-providers.
The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always UNREACHABLE.
I can just see in the log:
[May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
2005 May 17
4
multiple sip accounts from same sip registrar
Dear all,
I have an asterisk sip issue which I don't believe is unique.
I use a registrar (sipgate.co.uk) where I have 3 different accounts.
These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users.
By using just one of these accounts I can set asterisk up to send and receive calls no problem.
However, when I start to introduce an
2004 Jun 15
3
Grandstreams randomly go busy with Asterisk?
I've searched the lists but I didn't find anything exactly like this.
I have two Grandstream BT101 phones connected to an Asterisk.
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy. Dialing that
phone gives:
-- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack
2005 Mar 21
1
Net2Phone / Vonage
I can cut and paste the log file from a reload right now, and provide
you with the other information when I get home after work:
tmp*CLI> sip debug
SIP Debugging Enabled
tmp*CLI> reload
Mar 21 14:52:42 NOTICE[23231]: indications.c:397
ast_unregister_indication_country: Removed default indication country 'us'
11 headers, 0 lines
Reliably Transmitting:
REGISTER
2004 Jul 08
2
Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The
asterisk box is on a WAN connection on the other end of a DS3, the
phones connect fine to the Asterisk server as you can see from the
output of show sip peers below.
tp3/tp3 <firewall-ip> D N 255.255.255.255 60665
Unmonitored
tp2/tp2 <firewall-ip> D N
2007 Feb 03
3
error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
The following strange conditions is happening while I try to dial a
SIP user from another SIp user.
SIP to Zap dialing is fine, as all 4 users can call PSTN.
I'm using Asterisk SVN-branch-1.2-r51359M
Example: extension 3210 calls extension 3213. They are all registered properly:
chrom01*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
3213/3213
2005 May 12
1
realtime sip show peers no nat
Hello
sip show peers does not mark hosts as NAT even though sip.conf and
sip_peers table has nat=yes.
spitfire*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask
Port Status
voipuser.org/gdsm 216.127.66.119 N 255.255.255.255
5060 Unmonitored
5560/5560 192.168.4.5 D N A 255.255.255.255
5060
2010 Jun 15
4
can't seem to register, status unmonitored
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
<sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013
208 is ip of the asterisk server.
on the server on doing 'sip show peers' , it
2008 May 02
2
sip show peers
When doing a "sip show peers" I might see something like:
Name/username Host Dyn Nat ACL Port
Status
devcentos5x64_to_mmfirepa 192.168.1.177 5060
Unmonitored
devcentos5x64_to_bt610tMM 192.168.1.159 5060
Unmonitored
devcentos5x64_to_am2mm/de 192.168.1.178 5060
2005 Aug 02
2
asterisk@home newbie extensions always busy
hi list,
I'm running a newly installed asterisk@home an i registered two soft
phone. both soft phone are registered
8901/8901 x.x.x.x D 255.255.255.255 50710 Unmonitored
8900/8900 y.y.y.y D 255.255.255.255 6281
Unmonitored
but when I call one another, they are always busy and directed to its
voicemail
Sorry, if this was posted before
TIA
2011 May 12
2
Realtime - ara180
Hi all,
A week or so down the list, i read that not many people were using
realtime on an Asterisk18, so i had this afternoon a go at it...
[sorry for the inconveneant line-wraps]
First i did:
mysql> create database asterisk;
mysql> grant all on asterisk.* to 'voipadmin'@'localhost' identified by
next i used the info from the wiki:
CREATE TABLE `sip_devices` (
`id`