Hi,
I cannot get asterisk to start again after the g729 install failed.
kindly advise what's the problem.
Thank's
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Hello,
I have been tearing my hair out on this issue for 2 days, any help
would be appreciated.
We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch
There are two VLANs, 1(data) 50(VoIP). When Polycoms are connected
to the switch with VLAN 50 hard coded in the config
On 10/07/2010 06:50 PM, Daniel Tryba wrote:
On Thu, Oct 07, 2010 at 02:57:27PM +0200, Jonas Kellens wrote:
nat=yes is set as a global parameter and also in the realtime MySQL
sip_buddies database I have for every peer nat=yes.
I then find it very strange that when placing these Snom
Hello,
there is a really great difference in the Via-header of the
REGISTER-message between the Zoiper and the Snom.
Also the Zoiper has a Contact-header, and the Snom REGISTER has not...
Snom :
REGISTER sip:sip.domain.tld SIP/2.0
_*Via: SIP/2.0/UDP
On 7 Oct 2010, at 23:57, steve casto wrote:
A Crisco RVS4000 installed now has real problems with Sip, one-way audio and
throughput not up to the WAN speed.
ALG? (Assuming you mean Cisco..)
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Hi,
How do you typically test voice quality in Asterisk? For example if you like to
do load testing, or monitor voice quality and get notified if certain calls had
bad quality for proactive maintenance?
Thank you!
Best Regards,
Sevana Oy
http://www.sevana.fi--
The professional way is to do a series of test calls, play a reference
file and record the audio at the incoming side. You then use both files
to calculate a MOS score. This method is used by telco's to do quality
checks.
https://secure.wikimedia.org/wikipedia/en/wiki/Mean_opinion_score
On Fri, Oct 08, 2010 at 02:24:11PM +0200, Bert Van Kets wrote:
The professional way is to do a series of test calls, play a reference
file and record the audio at the incoming side. You then use both files
to calculate a MOS score. This method is used by telco's to do quality
checks.
Take a
Those boxes run around $50k USD, I've only seen them once back in the late
1990s.
At work for customer consulting we have very expensive site licenses for
Prognosis IPT Assessor which generate great looking pdf reports.
We also use Cisco IOS IP SLA however it doesn't have a reporting
You´re right!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Fri, 8 Oct 2010 00:16:58 -0500
From: sruff...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dahdi error
On 10/7/10 2:07 PM, Flavio Miranda wrote:
One quick clarification please... With Fluke ACEs you measure MOS according
G.107, E-model, right?
Thanks a lot to all who replied and will reply!
- Original Message -
From: Daniel Tryba dan...@tryba.nl
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
I'm trying to make a attended transfer through AMI. I though i could use
Atxfer, and it seems ok, but nothing happens.
And I can't find any how-to or description on how to do this. What more
do I have to do to make this work?
In Asterisk Call Manager:
Action: Atxfer
Channel: SIP/36-xx
I've hit an odd issue in a test 1.8 deployment,
playback() stalls mid file. The call stays up, but asterisk stops sending
packets.
It doesn't always happen - but on demo-congrats it happens about half the time.
It only happens in IAX calls.
Anyone else experienced it ?
(I filed an issue just
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton
Sent: Friday, October 08, 2010 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Weird stalling of playback
On 8 Oct 2010, at 15:37, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton
Sent: Friday, October 08, 2010 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
That switch doesn't seem to support CDP, so the Polycom phone has no way of
figuring out which VLAN to tag itself to automagically. It will grab the
primary VLAN unless you specify otherwise in the phone's setup.
On boot of the phone, go into setup, default password 456, there's an option in
Tim
I am actually seeing this on a 1.6.2.13 box as well. For some reason
durring prompt playbacks they some times stall mid file. The call stays up
but no audio comes in.
Bryant
From: Tim Panton t...@westhawk.co.uk
Sent: Friday, October 08, 2010 10:38
On 8 Oct 2010, at 15:37, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton
Sent: Friday, October 08, 2010 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 8 Oct 2010, at 16:03, Bryant Zimmerman wrote:
Tim
I am actually seeing this on a 1.6.2.13 box as well. For some reason durring
prompt playbacks they some times stall mid file. The call stays up but no
audio comes in.
Bryant
From: Tim Panton t...@westhawk.co.uk
Sent: Friday,
One more thing: Make sure that the port going to your data-DHCP server doesn't
have the voice VLAN set on it. I troubleshot an installation for a few hours
before thinking of this...
Bests,
Seb
On 2010-10-08, at 2:37 AM, Thermal Wetland wrote:
Hello,
I have been tearing my hair out on
On Thu, Oct 7, 2010 at 11:25 PM, Fazil Amaan fazil_d...@yahoo.com wrote:
Hi,
I cannot get asterisk to start again after the g729 install failed.
kindly advise what's the problem.
Thank's
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-- Bandwidth and
On Fri, Oct 8, 2010 at 5:16 AM, Sebastien Thomas li...@amplisys.ca wrote:
One more thing: Make sure that the port going to your data-DHCP server
doesn't have the voice VLAN set on it. I troubleshot an installation for a
few hours before thinking of this...
Interesting, the DHCP server for
Glad to hear it helped you Dennison.
VPN is such a confusing beast to lots of people I think and hence the
responses to this thread were all sort of work around and sometimes
off-topic. It's also not well documented or maybe the feature is not widely
used within the Asterisk community. I think it
Kyle,
Got an empty response from you. Were you intending to give your feedback?
Regards,
Bruce
On Wed, Oct 6, 2010 at 8:10 PM, Kyle Kienapfel doctor.w...@gmail.comwrote:
On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
This is such an annoying issue
He sent a two liner
On Fri, Oct 8, 2010 at 2:25 PM, bruce bruce bruceb...@gmail.com wrote:
Kyle,
Got an empty response from you. Were you intending to give your feedback?
Regards,
Bruce
Are you using stun?
http://en.wikipedia.org/wiki/Session_Traversal_Utilities_for_NAT
--
On 10/06/2010 02:50 PM, bruce bruce wrote:
Hi Guys,
This is such an annoying issue whenever it comes up. The sender and
receive always receive the source public IP no matter what in the IP
packets but then SIP packets go out with something like 192.168.0.20.
In this instance, a Bell
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of
the three perform well in all enviroments. Between stablity issues, T38 and
DTMF talkoff all three suffer some combination of issues.
I am looking at Patton and Innomedia. Has any one tried either brand and
what is your
On Fri, 8 Oct 2010, Bryant Zimmerman wrote:
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the
three perform well in all
enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer
some combination of issues.
I am looking at Patton and Innomedia.
Hi,
I was wondering if anyone stumbled upon the correct event in a sip
NOTIFY (from a SUBSCRIBE)
to make the BLF lamps on a Linksys/Cisco SPA9xx/5xx go yellow?
I'm trying to differentiate between On the Phone and DND with the BLF.
Thanks.
-- James
--
Us too. Tons of SPA2102's out there working fine!
On Fri, Oct 8, 2010 at 4:36 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Fri, 8 Oct 2010, Bryant Zimmerman wrote:
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of
the three perform well in all
enviroments.
Thanks for the feedback.
I said previously, Asterisk receives packets like extens...@192.168.0.10 is
trying to register to it. So, Asterisk sends out to local LAN an ACK which
obviously is not right. SPA-2102 should send SIP request like
extens...@123.123.123.123 (public IP).
Thanks
On Fri, Oct
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