Hi,
Where is that file? I am using Asterisknow 1.5. Please tell me the location
of the file
*
Thanks,
Arun S*
On Wed, Dec 30, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote:
How is DAHDI-1 set up in users.conf?
You need something like this
; Span 2: WCTDM/4 Wildcard TDM400P REV I
Hi,
It is not working. The same error and no CID is the result.
Thanks,
Arun S
On Wed, Dec 30, 2009 at 8:48 PM, Anthony Francis - Handy Networks LLC
anth...@handynetworks.com wrote:
You need to wait at least 1 second on an incoming POTS line for CID info,
add a wait(1) as the first step
[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid
extension '11', but no rule 'i' in context ...[snip]...
When testing IVR and pressing 1 from my Grandstream SIP-phone, the
above message is printed on the Asterisk CLI.
How come Asterisk receives my 1 as 11 ??
Settings in my
jonas kellens wrote:
[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid
extension '11', but no rule 'i' in context ...[snip]...
When testing IVR and pressing 1 from my Grandstream SIP-phone, the
above message is printed on the Asterisk CLI.
How come Asterisk receives my 1
Francesco,
Marking only RTP or only SIP info makes my DTMF to be correctly received
by Asterisk (read: only once).
It works fine now.
Thanks.
Jonas.
On Thu, 2009-12-31 at 10:53 +0100, Francesco Peeters wrote:
Jonas.
It may be me, but it looks like Asterisk correctly interprets the
hadi motamedi wrote:
Can you please let me know if we can have different codec schemes for
audio codec in audio codec out ? I mean , in one application , we
can have our audio codec input set to G.711 a-law and our audio codec
output set to G.711 u-law . I am facing with an application that
2009/12/29 VinÃcius Fontes vinic...@canall.com.br:
Hello everyone.
I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing.
Voice is working great, but I never configured anything using T.38 in
Asterisk so I'm kinda lost.
So you're trying to use a SPA8000 to act as a
Thanks to Digium, the company, and to all of the fine people from
Digium who participate in the weekly VoIP Users Conference conference!
We will be live on Friday January 1, 2010 and there is also a reel
of recorded greetings from people around the world wishing the VoIP
Community a Happy New
Hi,
I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH. After
the second person
answers both channels are bridged together.
Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once a week))
asterisk
I have a working dialplan for our phone system with Mon-Fri, business
hours identification, etc. But what I'm lacking right now is support
for company holiday dates.
What I'd like to do is to create a database of these dates and just
update them as new years rollover.
I suspect others have
Perhaps make the dates a database entry? The fixed dates would stay the
same each year and you would adjust only the floating dates.
Or, there are really few holidays in the year. (Unless you are a government
or a bank) Simple intercept code in the dialplan would handle most
businesses.
Just
Myles Wakeham wrote:
I suspect others have done this sort of thing with Asterisk before, but
I've not found any resources so far.
exten =
317xxx,1,Gosub(holiday_check,s,1)
[holiday_check]
;
;* Break out current 2 digit month
Sorry wrong topic...
Hi,
I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH. After
the second person
answers both channels are bridged together.
Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch,
have pstn-5665, digest has
Joseph wrote:
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username
mismatch, have
I
installed an Elastix based system and changed it to work in Device-Mode since
there is a call center and users has to login.
As
requested, I made recording always to all the users.
The
problem is there are no links in the Monitoring reports to the calls and while
checking
On Thu, Dec 31, 2009 at 12:12:19PM -0800, Yuval Yogev wrote:
I
installed an Elastix based system and changed it to work in Device-Mode
That's FreePBX terminology.
since
there is a call center and users has to login.
As
requested, I made recording always to all the users.
The
problem is
On 12/31/09 13:06, Kevin P. Fleming wrote:
Joseph wrote:
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553
On 12/31/09 13:06, Kevin P. Fleming wrote:
Joseph wrote:
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553
When I configured AudioCodes MP-114 to MWI it keeps complaining bout
subscription without mailbox:
chan_sip.c:15450 handle_request_subscribe: Received SIP subscribe for peer
without mailbox: pstn-5665
chan_sip.c:15450 handle_request_subscribe: Received SIP subscribe for peer
without mailbox:
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