Hello Armin Schindler,
Am 2012-06-28 09:52:30, hacktest Du folgendes herunter:
On 27.06.2012 18:46, Michelle Konzack wrote:
Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?
Which PCI-ID is that?
I do not know, because I have not bougth it yet.
It is only named:
Eicon DIVA
Hi
Does anyone have any experience of connecting SIP phones to an asterisk
server through the 2701HGV router that BT supply with their Infinity
product?
Thanks in Advance
Ish
--
Ishfaq Malik i...@pack-net.co.uk
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has
Am 29.06.2012 11:38, schrieb CDR:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until
In article 4feccd0c.1020...@fivecats.org,
James Sharp ja...@fivecats.org wrote:
On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
and Voip3 has a Wildcard
Hi Guys
Has anyone seen Dahdi dropping incoming calls with Hangup cause 27?
It only drops whilst we are on the phone?
Its not every single call
Any ideas?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony
Has anyone seen Dahdi dropping incoming calls with Hangup cause 27?
Please don't hijack a thread. Start a new message with your question, since
it'll screw up message threading.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety,
This is from the documentation of Perl-AGI
$AGI-stream_file($filename, $digits, $offset)
Executes AGI Command STREAM FILE $filename $digits [$offset]
This command instructs Asterisk to play the given sound file and
listen for the given dtmf digits. The fileextension must not be used
in the
On 12-06-29 05:38 AM, CDR wrote:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits
Quoting Ioan Indreias indre...@gmail.com:
On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar
maill...@lightspeed.ca wrote:
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and
Voip3 has a Wildcard TE110P.
I've never seen this on incoming calls, only outgoing calls.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Colin
Sent: Friday, June 29, 2012 8:11 AM
To: Asterisk Users Mailing List - Non-Commercial
Quoting Tony Mountifield t...@softins.co.uk:
In article 4feccd0c.1020...@fivecats.org,
James Sharp ja...@fivecats.org wrote:
On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard
Quoting Tony Mountifield t...@softins.co.uk:
In article 4feccd0c.1020...@fivecats.org,
James Sharp ja...@fivecats.org wrote:
On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard
- Original Message -
Curiously enough, I can't do that at all on Voip3. Not span 3 of
course, because only span 1 should exist, but I can't execute pri
show spans either.
DING DING DING... we may have a winner. Do you have PRI support on that box,
meaning, did you also compile
Dale,
Sorry for taking so long to answer, I've been traveling.
Thanks so much for the suggestion, your solution worked perfectly. I'm not
sure why I didn't notice that the IAX trunk was working in the other direction.
Once again, thanks for your help.
Mitch
Date: Mon, 25 Jun 2012 05:44:37
On 12-06-29 11:40 AM, Tim Nelson wrote:
- Original Message -
Curiously enough, I can't do that at all on Voip3. Not span 3 of
course, because only span 1 should exist, but I can't execute pri
show spans either.
DING DING DING... we may have a winner. Do you have PRI support on that
Hi all,
after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my
VoIP provider because it says I'm trying to connect to port 55150
(that's what the call center guy told me)...but I'm not. In my sip I've
set port=5060, not 55150.
The strange thing is that the rport inside SIP
Please use more meaningful subjects.
'Please dont tell me this is impossible'
I'm sure there are lots of things I could tell you that are impossible.
'Issue with Perl-AGI'
I'm sure there are many issues with Perl and AGI if you don't understand
the protocol.
Better subjects = better
On 29/6/12 9:59 am, Ishfaq Malik wrote:
Does anyone have any experience of connecting SIP phones to an asterisk
server through the 2701HGV router that BT supply with their Infinity
product?
Good luck with that.
The BT 'Home Hub' and 'Business Hub' routers they supply with retail
ADSL and
Quoting Tim Nelson tnel...@rockbochs.com:
- Original Message -
Curiously enough, I can't do that at all on Voip3. Not span 3 of
course, because only span 1 should exist, but I can't execute pri
show spans either.
DING DING DING... we may have a winner. Do you have PRI support on
Has anyone seen Dahdi dropping incoming calls with Hangup cause 27?
It only drops whilst we are on the phone?
Its not every single call
Any ideas?
Libpri can generate that cause code when T309 expires. T309 starts
when the link goes down. When T309 expires, active calls are dropped
because
- Original Message -
Quoting Tim Nelson tnel...@rockbochs.com:
- Original Message -
Curiously enough, I can't do that at all on Voip3. Not span 3 of
course, because only span 1 should exist, but I can't execute pri
show spans either.
DING DING DING... we may have
Hi,
I work for a VoIP provider in Southern California. We are looking
for someone very knowledgeable in Asterisk/VoIP to help work on the following:
- Maintenance of current Asterisk servers, updating Asterisk, monitoring
load, and other sysadmin tasks
- Devise and implement scalability
voice mail folder, I saw a .lock file.
Apparently this was caused by a core dump in the mail module. I witnessed this
just a bit ago. There are core files in /tmp. I'll search Jira for outstanding
tickets this weekend and open one if not found.
Doug
--
Ben Franklin quote:
Those who
Doug:
You may want to apply the patch on ASTERISK-19923 - it fixes a critical
problem in app_voicemail in the latest version.
We are planning on releasing a new version of 1.8.13/10.5, which
will include this patch.
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW -
We've deoplyed a number of pure VoIP wireless (wifi proprietary) phones, but
not dect.
Is there a simple overview of integrating DECT phones with Asterisk somewhere?
I assume the DECT basestation has a multi-account SIP VoIP interface, and the
handsets are just plain old dect?
Can you push
You may want to apply the patch on ASTERISK-19923 - it fixes a critical
Thank you for the info, I'll apply it this weekend!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
On Fri, Jun 29, 2012 at 1:22 PM, Michelle Dupuis mdup...@ocg.ca wrote:
We've deoplyed a number of pure VoIP wireless (wifi proprietary)
phones, but not dect.
Is there a simple overview of integrating DECT phones with Asterisk
somewhere? I assume the DECT basestation has a multi-account
Can you really mix match any base station with any DECT handset?
Do handsets have proprietary features which only work with their own
basestations? (eg: transfer between handsets)?
Can i buy a good base station and get cheap Costco Dect handsets?
From:
On 29/6/12 11:16 pm, Michelle Dupuis wrote:
Can you really mix match any base station with any DECT handset?
Yes and no.
Do handsets have proprietary features which only work with their own
basestations? (eg: transfer between handsets)?
Yes. And that's the 'no' part of my answer above -
I like the look of the C610H. Is there a matching DECT base station by Gigaset?
(I can't figure this out looking at their site)
I see a C610IP but it's not clear if that base station supports multiple SIP
accounts, multiple calls active.
From:
On 30/6/12 12:12 am, Michelle Dupuis wrote:
I like the look of the C610H. Is there a matching DECT base station by Gigaset?
I use the N300IP. Supports 3 active SIP calls I believe - and yes, does
have multiple SIP accounts (6, if I recall correctly).
Kind regards,
Chris
--
This email is
Do the C610H and C300IP use an international standard for frequencies? I can't
even find gigaset sold in USA/Canada...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
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