Hello,
I opened bug #19763 on jira last friday (20/04) and didn't get any
feedback till now. Is this a normal delay?
Regards
--
Daniel
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hi
I have a iax configuration:
[IaxServer]
type=friend
host=172.16.1.14
port=4569
defaultuser=IaxServer
auth=md5
secret=mypassword
context=Internal
peercontext=Internal
qualify=yes
trunk=no
disallow=all
allow=alaw
i see the peer:
ipbx*CLI iax2 show peers
Name/UsernameHost
Am 24.04.12 09:27, schrieb Administrator TOOTAI:
Hello,
I opened bug #19763 on jira last friday (20/04) and didn't get any
feedback till now. Is this a normal delay?
Regards
Hi,
i didnt want to say that it is a normal delay but most bug marshals
and devs work on asterisk bugs in their
Hi Sammy,
Yes my telco have a lot of IP, i receive a call from ~20 ip ..
I can't put a subnet ?
best regards
Le 23 avril 2012 07:57, SamyGo govoi...@gmail.com a écrit :
Hi,
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
This line is telling you everything. The peer you've
I wonder if anyone from asterisk development can tell about putting a subet
in *host=192.168.2.0/26 *field.
I fear you may need to declare peers for those ~20 IPs in worst case.
On Tue, Apr 24, 2012 at 12:38 PM, Olivier CALVANO o.calv...@gmail.comwrote:
Hi Sammy,
Yes my telco have a lot of
On Tuesday 24 April 2012, Satria Anamarta wrote:
Thanks AJS :)
The different is only this:
21,22c21,22
rxgain=8.0
txgain=8.0
---
rxgain=0.0
txgain=0.0
29a30
That looks perfectly sane (it's quite normal to get an extra blank line added
on the end with certain editors).
Le 24/04/2012 09:56, SamyGo a écrit :
I wonder if anyone from asterisk development can tell about putting a
subet in *host=192.168.2.0/26 http://192.168.2.0/26 *field.
I fear you may need to declare peers for those ~20 IPs in worst case.
[MyTelco]
...
deny=0.0.0.0/0.0.0.0
Le 24/04/2012 09:37, Stefan Schmidt a écrit :
Am 24.04.12 09:27, schrieb Administrator TOOTAI:
Hello,
I opened bug #19763 on jira last friday (20/04) and didn't get any
feedback till now. Is this a normal delay?
Regards
Hi,
i didnt want to say that it is a normal delay but most bug
Hello,
is there a way to put a certain SIP peer on state busy ?
I know you can do this by pressing DND on your IP-phone, but can this
state also be set in the dialplan ?
Thanks.
Jonas.
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-- Bandwidth and Colocation
Hi,
Does anybody knows about the PSTN frequency parameter with on/off hook
times for the city Nicaragua. This is part of USA below to Mexico.
Regards.
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If it only match on host field, why are multiple permit field allowed? And
for what they are usable then?
Peers are matched against IPs in host field, however the permit/deny
fields restricts the peers in case host=dynamic. That's what I've learned
so far.
But for OP I think he definitely
- Original Message -
From: Administrator TOOTAI ad...@tootai.net
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 24, 2012 5:19:36 AM
Subject: Re: [asterisk-users] Question for a Jira bug marshal
Le 24/04/2012 09:37, Stefan Schmidt a écrit :
Am 24.04.12 09:27, schrieb
Hello,
Is there a way to make the the 'X' key next to the direction map close
the menu system or do anything? It's kind of annoying to have to press
the Line 1 key.
http://www1.digium.com/sites/default/files/support/d40_phoneusersheet.pdf
Thanks,
Dennis
--
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this
On Mon, Apr 23, 2012 at 09:53:08AM -0500, Kevin P. Fleming wrote:
On 04/21/2012 04:07 PM, bilal ghayyad wrote:
Dear;
The output of the ./configure that is related to dahdi is:
checking for DAHDI_RESET_COUNTERS in dahdi/user.h... yes
checking dahdi/tonezone.h usability... yes
checking
On Mon, Apr 23, 2012 at 02:36:03PM -0500, Danny Nicholas wrote:
Don't know about 1.8 but in 1.4 dahdi_genconf would update users.conf
Huh?
dahdi_genconf is part of dahdi. It also does not generate users.conf
(and does not generate a users.conf snippet by default).
which
could mess with
Not by default, but if you do dahdi_genconf users it will put a trunk entry
into users.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, April 24, 2012 10:31 AM
To:
I have an Asterisk server connected to a Nortel Pbx via an E1.
Everything works fine, I get calls in and out with callerid. The
problem that has been reported to me is the following scenario:
A call comes in from the PSTN and is answered by Asterisk. The person
dials the operator (1000)
Please post your E1 configs. If you are not using QSIG you should. On
the nortel side this only works well with R6.0 and later. I have a
simular setup but with Cisco UCM but the calls come into the Nortel
first and then can be passed back and forth between them with no problem.
On 04/24/2012
The E1 between the Asterisk and Nortel is using R2 for signalling. The
PSTN comes to Asterisk first and then send calls to the Nortel. When we
started we were just replacing an automatic operator/voicemail system
for the Nortel and all calls went there. The customer has been
gradually
You need to change it to QSIG or this will continue to be a problem.
On 04/24/2012 12:05 PM, Carlos Chavez wrote:
The E1 between the Asterisk and Nortel is using R2 for signalling. The
PSTN comes to Asterisk first and then send calls to the Nortel. When we
started we were just
I believe Voicepulse can do all of this.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Horn
Sent: Tuesday, April 24, 2012 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Looking
On 04/24/12 16:57, Russell Horn wrote:
I received an email today from Junction Networks that they are
substantially increasing their monthly fee to the point that I'd be
cheaper getting a line from my local phone company. I'm now looking
for a replacement US carrier that supports IAX2.
I'm a
Joseph wrote:
On 04/24/12 16:57, Russell Horn wrote:
I received an email today from Junction Networks that they are
substantially increasing their monthly fee to the point that I'd be
cheaper getting a line from my local phone company. I'm now looking
for a replacement US carrier that
Hi Carlos
How about to perform the call transfer operation (screened and unscreened) and
trace the communication it will better on Nortel PBX side, in order to know
which point is causing call transfer failure.
By the way these kind of features is complicated to handler on interconnected
On Tue, Apr 24, 2012 at 6:17 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Voip.ms is high quality, handles number ports and supports both IAX2 and SIP
2 different pricing plans, and their costs range from 4.95 to 7.95 per month
depending on the rate center for one plan, and less with
On Tue, 24 Apr 2012, Russell Horn wrote:
On Tue, Apr 24, 2012 at 6:17 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Voip.ms is high quality, handles number ports and supports both IAX2 and SIP
2 different pricing plans, and their costs range from 4.95 to 7.95 per month
depending on the
Hi all:
I want to modify Dialplan or chan_sip.c to let asterisk server send some
abnormity messages to test mobile
such as:
3xx 302 Moved Temporarily Temporary redirect
4xx 400 Bad Request Indicates request error
401 UnauthorizedIndicates that authentication
Hi
No idea ?
thanks
Olivier
Le 24 avril 2012 16:06, Olivier CALVANO o.calv...@gmail.com a écrit :
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you
are trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a
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