Displaying 20 results from an estimated 5000 matches similar to: "Integrating Asterisk 1.8 with Google Talk and Google Voice"
2010 Oct 25
4
google voice + asterisk: calls made to GV# processed but weird
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
2008 Oct 15
1
Cisco 7960 not always receiving incoming calls
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest
2008 Oct 19
6
adding a second extension
I'm trying to add a second extension to my setup. The second device is
able to successfully connect to the Asterisk server. I am unable to
contact extension 101 from 102 and vise-versa. Also are my context
setup logically or is there a better fashion to organize them? My
error is at the bottom.
Here is the extension.conf
[default]
;
; By default we include the demo. In a production system,
2006 May 31
9
Unable to use 'valid users' from Active Directory
I am able to return users and groups using wbinfo -g and -u. Samaba will
even allow users to connect that are in our domain. The problem exist
while trying to narrow down permissions to a share.
[public]
comment = Public Stuff
path = /home/
public = yes
read only = no
valid users = @"UFAD\_IFAS-FRE-USERS_autoGS"
This does not work. It prompts the end user for a
2008 Oct 04
5
Vitelity Asterisk configuration help
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup incoming and outgoing calls with voicemail
support. I've searched all over but many of the
2008 Oct 19
2
Latency woes, qos the fix?
My latency is kind of high and the voice delay is noticeable.
The Asterisk server is on a dedicated host outside of the network. I
am performing PAT/NAT using a Cisco router.
ns1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.116
2014 Dec 25
2
originate , callerid
25.12.2014 15:46, Anthony Messina ?????:
> On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote:
>> I want to change call files, which has caller id in them, to call
>> originate from dial plan.
>> But I don't see such parameter here
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate
>>
>> How can I pass callerid
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2008 Aug 21
1
OT - Asterisk-Stats - Billsec instead of Duration
Hi,
To check telco billing, I'm usinfg Asterisk-Stats from
http://www.areski.net/asterisk-stat-v2/about.php .
How can you tweak this application to display graphics and data that use
Billsec instead of Duration ?
Regards
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2007 Oct 19
1
FollowMe recorded name filename variable?
Is there a variable for the filename that is created by the FollowMe
application when "a" is specified as an option to record the caller's name?
I'd like to clean up the recorded name files after the call is complete.
Thanks -Anthony
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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2007 Dec 05
3
No timezone in Voicemail email?
Hello,
I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out
when a user gets a voicemail don't have the timezone set in the header, so
they're appearing in the user's email clients at the wrong time. Has anyone
else seen this? I didn't find any bug reports or other info with Google.
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road,
2008 Oct 24
2
Asterisk and Cisco Call Manager Express (CME)
I was thinking about complicating my Voip setup by adding CME. I found
this example here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
and here: http://www.pasewaldt.com/cme/cme_index.htm
Would anyone like to comment on their experiences using CME with Asterisk...
I would like one of my Cisco phones to remain SIP connected directly
to my Asterisk system. The
2008 Feb 25
4
TDM400P dialout problem
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing
out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3.
I get the following:
-- Starting simple switch on 'Zap/1-1'
-- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack
[Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing
2017 Jul 30
2
dahdi kernel module
Does anyone know if there are any plans to update the dahdi-linux kernel
module code? It no longer compiles with recent kernels, and the last
release of dahdi-linux appears to have been around March of 2016. I am
currently running 4.6.3-300.fc24.x86_64 (on a Fedora system obviously) and
the dahdi-linux-complete-2.11.1+2.11.1 release builds and runs under this
kernel, but if I try to build it under
2008 Dec 20
2
Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far:
[incoming-fax]
exten =>
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM})
exten => s,2,ReceiveFAX(${FAXFILE}.tif)
exten => s,3,Hangup()
exten=>h,1,System(/usr/local/bin/fax2mail --cid-number "0${CALLERIDNUM}"
--cid-name "home fax"
2013 Jan 28
3
RPM updates
Hi All,
Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to..
Cheers
Steve
2014 Dec 25
3
originate , callerid
Hello!
I want to change call files, which has caller id in them, to call
originate from dial plan.
But I don't see such parameter here
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate
How can I pass callerid to following:
exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x")
Thank you!
2014 Mar 29
1
Unable to build DAHDI-Linux in mock chroot
Unfortunately, after
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398
I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I
believe this is related to the Makefile calling install_firmware with only 2
args, where install_firmware is a shell script with DESTDIR set to $3, which
is empty.
In this case, the DESTDIR
2011 Apr 27
1
Echocancellation OSLEC vs MG2 ?
Hi All,
Which echo cancellation is good between OSLEC and MG2. Dahdi by default use MG2 echo cancellation on channel. If i want to use OSLEC then what should i need to do ? Do i need to recompile dahdi with OSLEC ?
-S
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2009 Mar 10
1
Phone Directories/Asterisk/SIP/directory.html
Greetings!
We are using cisco 7940 phone with SIP and asterisk. We would like to be
able to have phone directories available on the phones that are sourced from
active directory. Are their any scripts that can connect to the AD server
via LDAP and then create the directory.html file for the phones?
Thanks!
Liz
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