Displaying 20 results from an estimated 28 matches for "telecommatters".
2006 Jun 12
1
MOH too loud
I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is
there a simple way to reduce the gain without having to remix the tracks?
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Jan 20
5
Asterisk in SPA9000?
Did Linksys really use Asterisk for the SPA9000 software?
--
Andres
Technical Support
http://www.telesip.net
2006 Mar 05
6
Polycom 501 power over ethernet
When I bought two Polycom 501 SIP phones, I naively thought they were
Power-over-Ethernet (IEEE 802.3af) because they were "powered over
ethernet." Silly me.
Polycom must have some odd voltage or funny way of injecting the
power, because the POE switch I bought for them (Netgear F@510P)
won't power them, though if I use the Polycom-supplied AC adapter and
ethernet power
2006 Feb 23
3
GPS-enabled cell phone/PDA
...ation is being monitored, so privacy rights are not an issue. The
subject will make periodic calls to the Asterisk server in order to
record his movements.
Does anyone have experience in this area?
Thanks,
Mike
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Mar 30
9
How is Teliax ?
Hi
I am looking at purchasing some DID lines from Teliax to install it on my
asterisk.
i would like to know some feed back on "Teliax" before i purchase.
suggest me if there are better sevice providers.
thanks
Giridhar Bandi
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2005 May 26
1
deadlock
All out of the blue I get these errors?
Any Ideas why
Please help
May 26 09:54:28 WARNING[3964]: channel.c:507 ast_channel_walk_locked:
Avoided initial deadlock for 'SIP/301-b9d0', 10 retries!
May 26 09:54:30 WARNING[3964]: channel.c:507 ast_channel_walk_locked:
Avoided initial deadlock for 'SIP/301-b9d0', 10 retries!
May 26 09:54:33 WARNING[3964]: channel.c:507
2006 Feb 24
1
Call quality problems
...e Polycom, each click can be
heard, the tone starts, but the tone is clipped and there is silence
until the next click.
I've verified that QoS is enabled in the IAD.
I would appreciate your thoughts.
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Mar 16
0
Testing IAX links
...opean server and just listen, but is there a more scientific method
to collect QoS metrics?
Thanks
P.S. I'm getting a lot of "Page Not Found" on lists.digium.com. Are
the older posts being purged?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Mar 23
1
SIP - Problem with audio clipping
...ith two different Asterisk system using the same
CLEC. One is * v1.2.4 and the other is v1.2.5. The systems have
totally different motherboards.
Has anyone had a similar problem, and what was the cause?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Mar 23
1
"Not Found" in archive
I'm seeing quite a few "Not Found" pages when I google lists.digium.com.
Is anyone else getting this?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Apr 01
1
Incorrect CDR results
...go out to
NuFone on IAX. Are these calls bridging away from the Asterisk server?
How can I get accurate billing data?
I tried to Google the archives but I'm still getting "page not found".
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Apr 04
0
Jitter in SIP calls?
...th the Asterisk system
or the telephones.
The Asterisk version is 1.2.5. The phones are Polycom, Cisco, and
Grandstream.
I've checked my NIC connections and everything is full duplex.
Thanks for your help.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Apr 04
0
Jitter in SIP connection
...ith the Asterisk system
or the telephones.
The Asterisk version is 1.2.5. The phones are Polycom, Cisco, and
Grandstream.
I've checked my NIC connections and everything is full duplex.
Thanks for your help.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Apr 14
2
Polycom 501 resource full problems ...
Hi List,
Not sure if this is the place for this so here goes ...
We have a number of Polycom 501's connected to our * box and they work
great. Some of our users have added a few entries into the directory on
the phone. The problem is on those particular phones they now sometimes
get "resource full" on the phone when accessing the directory. No central
directory was configured.
2006 Jun 24
2
Polycom 601 question
Hey everyone,
I know this isn't a direct Asterisk issue, but some of you may know this
answer.
I recently upgraded the SIP version to 1.6.6 on all of our phones in the
office. Everything is working fine, except one aspect. The phones in the
office reboot randomly for no apparent reason. I haven't changed
anything in the configuration files since the upgrade. The only setting
in the
2006 Oct 12
1
Attended transfer hanging PRI channel
...t.
The reason this is a big problem is that the PRI channel for the call
remains busy. Subsequent inbound calls on that channel are rejected.
Asterisk 1.2.12.1, Polycom SIP 1.6.6.
Has anyone seen this? Thanks.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 Nov 22
1
Recordings for VR analysis
Is there a programmatic to to trim the silence from the beginning and
end of a recording? From a .wav file? From a .ulaw file?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 May 15
1
Outgoing Calls Not Working all the time
I currently have Asterisk 1.2.7.1 and the Sangoma A200 w/ 6 FXO ports and HW
Echo canceller. I have outgoing calling setup to use a group so that if one
channel is busy it goes to one of the other channels. What's weird is that
when I dial an outside number, sometimes it goes through and other times I
get "You have reached an invalid pager number MCLL327." I have no idea what
that
2007 Mar 28
3
PoE - IEEE 802.3af
Hi,
I'm not clear on how to use Power--over-Ethernet, specifically with Polycom
phones.
What I understand, is that by buying the Polycom 501 with the 802.3af cable
bundle, I simply connect my phone, through the Polycom provided "special"
RJ-45 cable, into a PoE capable switch, and voil?!
Is this true? And if so, what happens when the Phone doesn't connect
directly to the
2006 Jan 04
5
Grandstream web configuration utility
I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386.
Browsing to each device by IP address, I can get logged in using admin
and I can see the advanced settings, however, if I try to change the
settings and clicking the Change button, it just brings me back to ask
for the password again..
I can't get into the Status page or any of the Account1-4 pages either.
It just keeps