Hi list,
This is comman issue when we use ODBC connection to get Database values. At
the 1st attempt connection is not connected but soon automatically connected
by aterisk. What is the real problem ? it this asterisk ODBC issue or else ?
-- Executing [_36899XX@incoming:1]
hi,
Dear all.
I setted a sip account on a sip trunk. when a client call via this sip
trunk, asterisk call failed on this trunk.
I have captured the sip messages on the host where asterisk located, and found
that:
1. asterisk send a INVITE message to remote sip proxy without
This is just a speculative shot in the dark, but remember that the domain of
the From URI is important, and that the authentication realm (domain) is part
of the authentication credentials. So, what you have in your 'fromdomain' and
'host' settings on the peer does matter.
--
This message was
What Sip declaration are you using for the remote sip proxy in sip.conf?
On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov
abalas...@evaristesys.comwrote:
This is just a speculative shot in the dark, but remember that the domain
of the From URI is important, and that the authentication realm
hi, alex
thanks for your kindly reply.
the remote proxy do not use domain as realm, the following is part of the
message
210.83.80.xxx is the ip of asterisk
202.104.188.xx is the ip of remote proxy sip server ( not asterisk)
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hi, Sam
thanks for your kindly reply.
The remote proxy is not asterisk
2011-09-30
cnasterisk
发件人: Sam Govind
发送时间: 2011-09-30 15:36:41
收件人: Asterisk Users Mailing List - Non-Commercial Discussion
抄送:
主题: Re: [asterisk-users] invite authentication error !?
What Sip declaration are
Whatever that remote party is, you are most definitely using
a username/secret declaration for that. So the sip attributes set for that
proxy define the behaviour for this.
On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk cnaster...@163.com wrote:
**
hi, Sam
thanks for your kindly reply.
The
On Friday 30 September 2011, NaJIm wrote:
Am I getting these error messages due to wrong configurations in my
zapata.conf. ??
I have got the following configurations in my zapata-channels.conf.
; Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED
group=0,11
context=from-pstn
asterisk can register successfully on the remote party. so i think username
password must be ok
2011-09-30
cnasterisk
发件人: Sam Govind
发送时间: 2011-09-30 16:05:21
收件人: Asterisk Users Mailing List - Non-Commercial Discussion
抄送:
主题: Re: [asterisk-users] invite authentication error !?
Please post some configurations. YOU CAN REMOVE username / secret from the
sip configs but posting remaining configs will only help to your issue
resolution, Alex first words were This is just a speculative shot in the
dark .
Fine if you were just Trolling here !
2011/9/30 cnasterisk
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme timing.
Doing core show translation give on the
Friends,
SIPit is an event organized by the SIP Forum and partners. It has been running
for 15 years twice a year, making sure that SIP clients and servers
interoperate. By testing, we also find issues with the myriad of RFCs in this
area and correct them. Testing interoperability is
Hi
Thanks everyone for your help and support all works perfectly
Best Regards
2011/9/29 salaheddine elharit salah.elharit...@gmail.com
ok thanks for your response i will try that and i will update you as soon
as i have any result
best regards
2011/9/29 A J Stiles
On 09/30/2011 03:56 AM, Administrator TOOTAI wrote:
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme
Le 30/09/2011 14:05, Kevin P. Fleming a écrit :
On 09/30/2011 03:56 AM, Administrator TOOTAI wrote:
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no
On 09/29/2011 12:37 PM, A J Stiles wrote:
On Thursday 29 September 2011, Lyle McKarns wrote:
Hello Asterisk List!
I have been asked to record calls from specific agents, and I am having
difficulty finding if this is possible, and if so, how exactly to do it.
Have a recorded context and an
Hi,
I think use of any Macro in queue can serve you well. Macro will be called
whenever the call is established to the agent. In that Macro check your
desired Agent and if condition matched trigger MixMonitor else do nothing.
Regards,
Gohar A.
From: asterisk-users-boun...@lists.digium.com
On 09/30/2011 07:49 AM, Administrator TOOTAI wrote:
Le 30/09/2011 14:05, Kevin P. Fleming a écrit :
On 09/30/2011 03:56 AM, Administrator TOOTAI wrote:
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS
Is there any reason not to run Asterisk on an Intel Atom board?
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In article 4e85d19f.4090...@digium.com,
Kevin P. Fleming kpflem...@digium.com wrote:
On 09/30/2011 07:49 AM, Administrator TOOTAI wrote:
Le 30/09/2011 14:05, Kevin P. Fleming a écrit :
On 09/30/2011 03:56 AM, Administrator TOOTAI wrote:
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2
I always use the recalc option to show translations, it seems to provide much
more accurate numbers.
Example: core show translation recalc 20
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield
On 09/30/2011 09:53 AM, Tony Mountifield wrote:
In article 4e85d19f.4090...@digium.com,
Kevin P. Fleming kpflem...@digium.com wrote:
This is why the output was changed to microseconds from milliseconds; in
the older version, the lowest number that should be shown was 1
millisecond, even if
Le 30/09/2011 16:59, Eric Wieling a écrit :
I always use the recalc option to show translations, it seems to provide much
more accurate numbers.
Example: core show translation recalc 20
Lenny kernel, new values, still 1000 microseconds between both directions
Translation times
Le 30/09/2011 17:02, Jason Parker a écrit :
On 09/30/2011 09:53 AM, Tony Mountifield wrote:
In article4e85d19f.4090...@digium.com,
Kevin P. Flemingkpflem...@digium.com wrote:
This is why the output was changed to microseconds from milliseconds; in
the older version, the lowest number that
Hello Everyone,
We have setup Asterisk HA, basically what we have is:
Virtual IP (for asterisk 1 and 2)192.168.2.6
asterisk1 192.168.2.110
asterisk2 192.168.2.111
mysql192.168.2.100
At 07:50 AM 9/30/2011, you wrote:
Is there any reason not to run Asterisk on an Intel Atom board?
Mine's been running that way for 3 years or so. 2 users 6 extensions,
SIP + 3 POTs lines with a TDM04.
Ira
--
_
--
Hello members,
I am looking for USA incoming DID which can be registered on softphone/IP
Phone/ Pap2 devices.
The DID will only be required to receive inbound calls and no outbound
calls.
Let me know your best per month prices/cost for the above.
Regards,
Amit Mehta
--
Hi,
As Eric mentioned I made my zapata.conf and zaptel.conf to match each
other.
My *zapata.conf *was
*group=1*
*switchtype=euroisdn*
*signalling=pri_cpe*
*callerid=asreceived*
*usecallerid=yes*
*cidsignalling=dtmf*
*cidstart=ring*
*context=TEST_EXTERNAL*
*channel=1-15*
*channel=17-24*
This
Google is your best friend when looking for this type of assistance my friend.
try callcentric vonage packet8 for reliable retail DIDs.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Sat, 1 Oct 2011 00:51:59
Go to ATT and ask for a BRI ISDN Line. Tell them that you want a
system access with a number block.
that would be typically 54448[0-9]
where your extensions are from 0-9
I don't know what protocol the americans are using, as I know the
americans made for sure their own thing which is known as
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