Displaying 20 results from an estimated 10000 matches similar to: "Asterisk Manager Interface notes"
2004 Aug 27
2
Someone please try MeetMe MOH with latest CVS and GS phone
I have today reported a bug with the latest channel.c (1.134) that
affects music-on-hold for the first user in a MeetMe room when calling
from a Grandstream BT102. The music is broken up about 5-10 times a
second. It doesn't happen when calling from Firefly. It is also fine
on both clients with 1.133 of channel.c. I am using the ALAW codec.
Mark at Digium can't reproduce the problem,
2005 Feb 16
1
Inter-asterisk conferencing delays - IAX2 configuration problem?
Hi
We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes.
All the other legs are PSTN (TE410P). The example configuration
Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme
The delay is between Slave box 1 and Slave box 2
The primary suspect is our iax configuration
2003 Apr 19
7
Call screening
I've set up asterisk with my X100P as a home answering machine. Works great
so far - answers the phone after 20 seconds, runs the phone tree, emails
voicemail, etc.
However, the one feature traditional answering machines have that I haven't
been able to figure out is how to listen in on the call. Ideally I could
just route through Console/dsp and hear it on my speakers. I've tried
2004 Feb 06
4
Conference server
Hi, we are setting a 120-channel conference server and would like to
learn if someone already did this (hardware, problems, etc...)
Best regards,
PauloHM
2005 May 25
1
Can Ztdummy be used in production environment
Hi there
I have been using Asterisk Meetme with Ztdummy for timing. It seems to work
fine and I haven't had any major problems. I am now moving into a production
environment and am wondering if it is better to use a Zaptel card? Are there
any problems with Ztdummy? I will probably have around 30 concurrent users
in different conferences.
Many thanks
Steven
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2003 Nov 15
10
MeetMe problem
Hi guys,
Having a bit of a problem trying to get conference bridges working. In my
meetme.conf file I have the following line
[rooms]
conf => 6000
In my extensions.conf file I have:
exten => 1000,1,MeetMe,6000
My problem is that when I dial into extension 1000 it is telling me "this
is not a valid conference number". Can anybody telling me what I'm doing
wrong here?
2008 Jul 22
2
3-way calling for IAX channels
How can I made a 3-way conference betwwen IAX channels?
My current version is: 1.4.21.1
Thanx,
Daniel Arohuanca Lagos
+51 1 3594122
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2007 Mar 11
4
Problem configuring voice conference
Hey!
I am trying to configure the voice onference with
MeetMe application for my internal users. I have my
server and 4 clients on same LAN and following is my
extensions.conf file:
[globals]
Ahsen=SIP/222
Tahami=SIP/444
Uzair=SIP/333
Wasif=SIP/555
[internal]
exten => 1234,1,Macro(voicemail,${Ahsen})
exten => 4321,1,Macro(voicemail,${Uzair})
exten => 5678,1,Macro(voicemail,${Tahami})
2005 Feb 28
2
Advanced FollowMe or Forwarding Application Suggestions
Our company is at the point now where we're almost ready to switch over to
an Asterisk server for a number of telephony applications.
There is one final application I've been trying hard to find to replace
something we already use with another provider. It's kind of an advanced
"FollowMe" application. (freedomvoice.com)
It works as follows:
1. An outside caller calls into
2011 Apr 26
1
Asterisk 1.6.2.18 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.18.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.18 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this
2011 Apr 26
1
Asterisk 1.6.2.18 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.18.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.18 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this
2014 Mar 21
3
Need more meetme users -- hitting some limit
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
CentOS 6.5.
The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.
The primary application will be bridging groups of users using meetme().
I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1
behaving a bit more like a production box -- bridging calls (box2).
The call
2003 Dec 01
7
Call Announcement - How To ...
All,
I would like to play an announcement to the user on what external line a
call came in, right before this call get bridged to this user. How would I
go about implementing this in * ?
Regards,
Hans
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2007 Sep 26
3
How to "busy out" zap channels
I know this topic came up many months back and some discussions were being
had on how to do this within the Zaptel drivers. However, I'm looking for
even a crude hack that someone has put together to get this done.
We have PRI's and LD T1's that are load balanced on two boxes. The hunt
order goes from box to box as far as the spans are concerned. There are
times that I would like to
2004 May 24
5
2 Sip phones behind un-natted Asterisk
I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a
Linksys firewall that supports UPnP. The Asterisk server has a public
IP. Here are the problems that I am having with this configuration...
1. The 2 SIP phones can call MeetMe and have a conference but
cannot call each other. (Yes, they connect but no audio either
direction)
2. I have verify=yes in the sip.conf for both
2005 Feb 16
3
Monitoring Conferences
I have benn having trouble with the Monitor Command.
Basically any time that I send a call into a MeetMe room I am only able
to monitor half of the conversation.
File-in is recorded with the incoming voice but file-out does NOT record
anything.
I have tried this with both the b and m option as well as without any
options to the MeetMe command.
Also the Monitor correctly records both sides of the
2004 Sep 05
3
ChanSpy by anthm and more...
Everyone we have a few new things to give back to the asterisk community.
http://bugs.digium.com/bug_view_page.php?bug_id=0002379
http://bugs.digium.com/bug_view_page.php?bug_id=0002380
http://bugs.digium.com/bug_view_page.php?bug_id=0002381
These include app_chanspy, the ability to spy on ANY bridged call taking
place inside asterisk. NOT just ZAP as with ZapScan/Barge.
Native format_* files
2009 Dec 24
1
How to create MeetME room with dialplan?
Hi,
Is it possible to create a meet me room on the go through dial plan? I am
looking to use AMI Originate to drop a call into meetme room and once it's
proved that party is joined, play him an announcement, grab few numbers from
them, and then dial a second number and drop into the same meetme room. The
reason to use this is to be able to know when the channels connected because
both
2005 Mar 02
3
Asterisk Manager API - multi "Originate" cal ls
Hello,
You can do either, you can send multiple Originate actions in a long line
without waiting for a response back(although the responses do usually come
back very fast) or you can open multiple connections using each one to
Originate a new call. We use the multiple connection method in the
astGUIclient suite because if you get a pause or lag in Manager output on a
single connection(which does
2004 May 25
2
sip phone problem
Hi all.
I have 2 ip phones (Grandstream Budgetone):
-budgetone1
-budgetone2
All two are connected to an Asterisk server.
When I make a call from budgetone1 to budgetone2, I
can speak with budgetone2 whith no problem. But when
budgetone2 hangs up, budgetone1 does not play any tone
(like busy tone). Budgetone1 seems to be still in
conversation, but what conversation!
Has anyone had a problem