Looking for IP Phone recommendation to be used with FreePBX. Approx. 50 IP
Phones, the FreePBX behind router. So good NAT support needed. FreePBX is
installed from the iso image available in AsteriskNOW website.
Sorry for the cross post, hoping that the 2 lists provide 2 different
perspective.
Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly
NR, and I would reregister them from the phone web interface. But it started
doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code: snom190-SIP 3.56m
I am trying to get a SNOM 320 working with Asierisk.
It does register and I can make outbound calls. But it
would not take inbound calls. This is what I get;
-- Executing Dial(Zap/2-1, SIP/102|20|Tt) in
new stack
-- Called 102
-- Got SIP response 404 Not Found back from
192.168.1.105
Is it possible to invite a 3rd party into to
conference? Something like, conference is ongoing,
pressing # would allow to dial the number, if that
number answers, will be automatically added in the
conference.
I did search the mailing list, wiki pages and google.
There were some discussion but no
I am not sure if it is the right list for the post.
Please excuse my lack of expertise, if it is a bad
post.
Is there anyway to detect the originating network
identity of the call in Asterisk? For example, if the
Asterisk gets a call from Cingular Network, is there
anyway to find out that the
There are 4 options in ASTCC routes, when I go to edit
a route. How does that work?
Thanks
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Mike,
Please elaborate it little bit. I am having the same
problem. Are you using read() for the EXTEN variable?
if so, how?
Thanks
I found it, read() does exactly what I need
On Sun, 7 Nov 2004 06:09:51 -0800, Mike Roberts
manipura at gmail.com wrote:
I'm trying to do this from PSTN - DID
What should be the EMail Program in astcc-admin's
Configure page?
I tried sendmail, it did not send any mail to the
Admin Email address.
Thanks
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I am using Uniden UIP200 SIP Phone. While I was
talking in one line, another call came in. I tried the
to put the first call on hold. It would not put the
call on hold. But I could switch between the lines
with Flash.
When there is one call, the hold works fine.
Has anybody else found this
I bought a Cisco 7940, I need to configure it for
Asterisk. I checked the wiki pages. Followed the link
to Cisco web page. Tried to download the image for
SIP. It wo'nt allow me even though I registered for
the CCO Valet. Is the image available anywhere else?
I saw some of the messages in the
Kurt,
Thanks for your reply.
As I mentioned before I could not download the image.
Did you download it from Cisco's website?
Also how did you provision the phone, I would really
apreciate if you could detail the procedure.
Thanks
--- Kurt [EMAIL PROTECTED] wrote:
What I had to do was to
Prepaid app can not connect to the database,
[app_prepaid.so] = (Prepaid Application)
== Parsing '/etc/asterisk/prepaid.conf': Found
Jun 22 14:38:43 ERROR[-1084964736]: app_prepaid.c:127
check_connected: app_prepaid: cannot connect to
database server localhost. Calls will not be logged
==
Jeremy,
Please explain little bit more how to use the
extension logic for a prepaid app.
Example would be greatly appreciated.
Thanks
--- Jeremy McNamara [EMAIL PROTECTED] wrote:
Why not just use extension logic? (which is far more
powerful than
app_prepaid can ever dream of being)
I am trying to install the Modified Prepaid App. I
have installed PostgeSQL, created the tables, etc.
Make Install runs ok. The when I try to launch
asterisk (asterisk -vgc), it fails to run. I get
the following errors,
1st error:
[app_prepaid.so] = (Prepaid Application)
== Parsing
I want to buy a 7940 to use with Asterisk. Does all
the features (i.e. Transfer, Hold, call waiting, MWI,
etc)work?
How difficult it is to configure 7940?
Thanks
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