Displaying 20 results from an estimated 500 matches similar to: "adding a second extension"
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 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 09
2
Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => 101,1,Dial(SIP/101,20)
exten => 101,n,Voicemail(101 at default)
;This automatically
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
2010 Dec 22
1
Simplifying dial-plan
Is there a way to include:
_NXXNXXXXXX
_NXXXXXX
_011.
_911
into my current plan:
2010 Dec 19
2
Specifying DID for outbound calls
The host I am working with has two accounts from the same DID
provider. Incoming calls work correctly and dial the appropriate
extensions. This also allows incoming calls to be billed appropriately to the
individual DID accounts.
Outgoing calls from either extension default to the first DID, i.e.
calls from either extension have the same callerID. How can an
extension specify separate outgoing
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
2010 Oct 24
5
Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
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
2009 Nov 07
1
Trouble registering Cisco 7942
I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960
and 7912 currently connected and functioning. I'm trying to use the
recommendations from here:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
I have created a "XMLDefault.cnf.xml" and it took the latest image but
the phone states it's unprovisioned? Any
2009 Oct 16
5
IVR
Ladies and Gentlemen,
We already have an Asterisk Call center suite installed at our contact
center. Now we wish to commence IVR services. We are offering Health
Information Services. Can someone help us to develop this Addon / Solution?
Best regards.
--
?????? ????? ????? ???? ???????
Nazir Ahmed Vaid
Cell:+92300-8277778
eHealth Services (Pvt) Ltd.
http://www.ehealth-services.com
NexSource
2009 Oct 09
2
Incoming extension not working.
Hi, all. I'm probably doing Something Dumb(tm), so please feel free to
point out whatever I'm missing, no matter how stupid.
Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get:
Rejected connect attempt from 64.2.142.19, who was trying to reach
'6031234567@'
This leads me to my first question -- why doesn't it show a context?
(My second is,
2009 Jan 15
2
Has anyone used FaxGateway()
Hi,
I've been trying to use the FaxGateway application to send T.38 out
over Zaptel using asterisk but I don't seem to be having any luck.
I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number])
Has anyone had any luck using this thing and can enlighten me on how
it's supposed to be used?
Thanks.
2006 Aug 24
1
Strange permissions problems
I had this problem some last year and never got it figured out. Now it
is bugging me. It seems that sometimes when a student writes his/her
file to a directory, it will not keep the correct group. It puts
his/her main group as the group owner and that fouls things up. Here is
what I have.
Unix Permissions
/school 3777 admin.teacher
/school/bhs 3777 admin.teacher
/school/bhs/reese
2007 Apr 17
5
Session problem mongrel behind Apache proxy
Hi,
I''ve configured mongrel_clusters behind an Apache 2.2 proxy using
named virtual host. Session are saved as ActiveRecordSession. But the
cookies created on client side doesn''t correspond to session data
saved in database (keys are different). The RoR app react just like
it doesn''t have a session at all.
If I don''t use Apache as a proxy/load balancer
2008 Jun 03
3
Asterisk 1.4.20.1 with bad gsm file playback
Hi All,
I'm stumped on this and I looking for some clues to fix this.
This is a new install of Slackware 12.1 onto an IBM x330 Server.
Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just
fine, but when I play the gsm files the audio quite choppy. And, the files
produced from the MixMonitor don't even record any audio other than noise.
I have a hard drive from
2007 Aug 20
5
byte-range requests
Hello everyone,
I did some initial tests and it seems that mongrel does not support
byte-range requests. Is this correct?
The reason I ask is that the iPhone requires byte-range requests to
work in order to stream audio or video from a web server.
Thanks in advance,
alan
2011 Aug 09
2
[LLVMdev] Adding a module in a pass
On 2011-08-09 16:48, John Criswell wrote:
> On 8/9/11 6:49 AM, Bjorn Reese wrote:
>> I have an optimization pass (FunctionPass) where I need to add global
>> constructors.
>>
>> For cleaness sake I decided to add these in my own module. My module
>> is created in my FunctionPass constructor:
>
> This is not how I would do it. A FunctionPass has
2005 Jun 08
2
format g729 and Voxee.com
Hi,
I have just signed up with Voxee.com and have attached my Asterisk
server to dial them via IAX2.
Below is the start of the log which dials the number and promply
hangs up when the call is answered, with the logs saying that the
channel is not compatiable.
I have traced this down to the g.729 codec which I don't have
installed. Any ideas on how to force that the codec not be used?
2008 Sep 19
2
Dropping Phone Calls
Hi All,
I'm currently having trouble with dropped phone calls. The following error
message is always in the log. This is a Grandstream GXP-2000 Firmware
1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been
occurring on other versions also.
[Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum
retries exceeded on transmission 8acaea6dc4c6e9b5 at