Dear All
I have an application that calls for Asterisk sip configuration to be able
to communicate with external sip server . My Asterisk 3.1.14 has been
installed on Debian 3.1 server and the external sip server is
@192.168.0.10, the same subnet as my Debian server @
192.168.0.2 . At now , the
Hi all!
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
type=friend
regexten=0317998975
secret=
username=0317998975
callerid=Magnus Benngard
mailbox=0317998...@inputinterior.se
host=dynamic
canreinvite=yes
dtmfmode=rfc2833
nat=yes
disallow=all
John Regal wrote:
Hi All,
My client makes manual sales calls to prospects. He is often sent to
voicemail on the prospect's side. If he finds himself having to leave
a message, he would like to be able to press a key and let a
pre-recorded message play into the prospect's vmail box. This is
Magnus Benngård schrieb:
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
Set
call-limit=10
(or any other value 0)
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,NoOp(0317998...@inputinterior.se has state
Hello List.
I would like to know how I can use two or more service providers with asterisk
to be used randomly for ei, if an user tries to make a call I would like to
randomly use a provider. It doesn't matter where the call is destined to.
Thanks.
Since you seem to have reservations about all the hardware options
available, how about avoiding the problem altogether? Order remote call
forwarding on your POTS lines and call forward them to DIDs provided by
an ITSP. No FXOs required.
If there's a problem you defeat the call forwarding and
Hi,
My Asterisk systems runs like a dream with mISDN, SIP and even and old
Digium board. But have almost in every conversation some irritating DTMF
being generated. The seems to be just as often from all trunks but are
worse if noise load speaker in other end.
Any good advices?
Where to look
I have seen this years ago, i received complaints about women voices
triggering dtmf.
With some help from Mr. Underwood, it was able to confirm lots of false
positives on the dtmf detection.
My issues went away when we upgraded all cards to the ones with the
octasic DSP chip on them.
Zoa
On
That worked beautifully, Thank you VERY much.
John
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Saturday, December 12, 2009 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial
Does anyone have a script that performs Auto Attendant / Receptionist system
If so, please send.
Thanks
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I used to have this problem with a Digium 400p -- even when not in use
and a Motherboard which was inadequate in terms of the interrupts for
the Digium card -- when I got a better Motherboard the problem went
away.
hbk fo...@online.no wrote:
Hi,
My Asterisk systems runs like a dream with
Thomas Perron wrote:
Does anyone have a script that performs Auto Attendant / Receptionist system
If so, please send.
You need to be more specific. What are you looking for this to do?
Your question is too generic.
Doug
--
Ben Franklin quote:
Those who would give up Essential
I want to list 100 indiviual businesses.
and do an ivr for them specifically
some use databases so i need an agi script in .pl or php.
On Sat, Dec 12, 2009 at 7:26 PM, Doug Lytle supp...@drdos.info wrote:
Thomas Perron wrote:
Does anyone have a script that performs Auto Attendant /
Try: BackGround(es/good)
--
Sent from mobile device
On Dec 12, 2009, at 9:50 PM, Landy Landy landysacco...@yahoo.com
wrote:
Hi List.
Don't know if I already posted about this problem but, if I have I
apologize for the double post.
I am trying to test a time of day extension dialing
Hi List.
Don't know if I already posted about this problem but, if I have I apologize
for the double post.
I am trying to test a time of day extension dialing 80, all I'm trying to test
is if is morning I would like asterisk to say Good Morning but, when I run
the test I get the following
Same thing:
== Using SIP RTP CoS mark 5
-- Executing [...@outbound:1] Answer(SIP/102-096a48c8, ) in new stack
-- Executing [...@outbound:2] Verbose(SIP/102-096a48c8, In timeofday
) in new stack
In timeofday
-- Executing [...@outbound:3] GotoIfTime(SIP/102-096a48c8,
Take the whitespace out of your ()'s. It's:
exten = 80,n,BackGround(es/good)
not
exten = 80,n,BackGround( es/good )
Thanks,
--Warren Selby
On Dec 12, 2009, at 9:16 PM, Landy Landy landysacco...@yahoo.com
wrote:
Same thing:
== Using SIP RTP CoS mark 5
-- Executing
On Dec 12, 2009, at 9:16 PM, Landy Landy landysacco...@yahoo.com
wrote:
-- Executing [...@outbound:17] BackGround(SIP/102-096a48c8, es/
good ) in new stack
[Dec 12 23:24:07] WARNING[6343]: file.c:650 ast_openstream_full:
File es/good does not exist in any format
[Dec 12 23:24:07]
On 12/12/2009 10:59 PM, Steve Edwards wrote:
A perfect example of Asterisk's asinine handling of whitespace.
Either that, or a perfect example of user imprecision. It's OK to
demand a certain grammar from the users of what amounts to a scripting
language, including one that is sensitive to
On 12/12/2009 11:54 PM, John Novack wrote:
Alex Balashov wrote:
On 12/12/2009 10:59 PM, Steve Edwards wrote:
A perfect example of Asterisk's asinine handling of whitespace.
Either that, or a perfect example of user imprecision. It's OK to demand a
certain grammar from the users of what
Alex Balashov wrote:
On 12/12/2009 10:59 PM, Steve Edwards wrote:
A perfect example of Asterisk's asinine handling of whitespace.
Either that, or a perfect example of user imprecision. It's OK to demand a
certain grammar from the users of what amounts to a scripting language,
Hopefully un-top-posting correctly...
On 12/12/2009 10:59 PM, Steve Edwards wrote:
A perfect example of Asterisk's asinine handling of whitespace.
On Sat, 12 Dec 2009, Alex Balashov wrote:
Either that, or a perfect example of user imprecision. It's OK to
demand a certain grammar from the
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