Displaying 20 results from an estimated 10000 matches similar to: "Short Audio Drop Out During Calls"
2006 Jun 06
1
SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com
Dear list (and more specifically Bret),
I am getting one-way (inbound only) audio when trying to place a SIP call
via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out
"== Forcing Marker bit, because SSRC has changed" 5 times after atempting a
native bridge. I realize this is most certainly a NAT issue, the * server is
behind one. Sip.conf has externip=, and
2006 May 15
3
How to tell if RTP stream is has been reinvited?
Howdy,
How can you tell if RTP traffic has been reinvited/is bypassing an * server?
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
2006 Feb 27
2
Echo on PRI/BRI?
Howdy:
Does echo only occur on analogue PSTN lines, or can it also occur on PRI and
BRI lines? If so, for the same reasons? This is a part of our consideration
to transition to BRI.
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com
2006 Jan 30
1
Need to recompile * after changing zap echo method?
Dearest List,
I guess I missed this point: Is it true that if you change the echo canceler
in zconfig.h, and then recompile/install your zap modules, that for this to
be taken into effect by * you must then recompile/install *?
I would have figured that the zap echo cancellation method was independent
of *, and I don't recall seeing any docs mentioning either way.
Sincerely,
Brent A.
2006 May 31
1
Can you dial with different CID's?
Is it possible to dial more than one extension with a different CID to each
extension? I'm thinking macros might be needed, but I don't have a good
handle on macros. Is it possible? Any hints?
BTW - this would be used for showing an internal extension to one phone and
a PSTN accessible number to another phone.
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
2007 Jul 05
2
REGEX expression for NXXNXXXXXX?
Hola,
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXXXXXX?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:brent.torrenga at torrenga.com
web:www.torrenga.com
2006 Dec 18
1
Cisco 7940 - NAT Option
I am thinking of turning on the NAT option in our Cisco phones (and the
corresponding sip.conf modification) to allow the phones to be taken outside
the LAN.
Can anyone think of any reason not to just always turn on the NAT enabled
option? I can't think of a reason not to always operate these phones with
this enabled, since it would likely allow them to be taken outside our LAN
and used.
2006 Jan 27
6
Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.
Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server would reply to a
ping, but no ssh login, no local console login - just locked up. This ain't
good for
2006 Jun 14
3
Directory - First Name/Last Name - How to use both? a@h?
I think A@Home allows a user to search a directory by either first OR last
name, right? I don't know for sure since I don't run A@Home.
I would like to offer that functionality in my system - and I'd have done it
by now if there was a prompt where Allison asks "press 1 to search by first
name, press 2 to search by last name". But I don't think that prompt exists.
Can
2006 Jan 31
7
Teliax - Codec Preference effective?
Has anyone had problems getting their preffered codecs on the Teliax web
interface taking effect?
I have two accounts, two separate yet similarly configured * servers. On one
account the settings took right away - on another server I am getting no
result. In fact, no matter what I change the settings to, only the old
codecs are usable (otherwise * says it can't negotiate a codec). Teliax
2006 Feb 15
9
Random Disconnects - or ARE they?
I have one use on our PBX who has been experiencing seemingly random
disconnects. The user is on the same LAN as everyone else, using the same
type of phone (79XX loaded with SIP firmware) as everyone else. He had some
disconnects a few weeks ago, I suspected the phone, so I swapped his with
mine. I have since not had issues with his old phone, however, he has had
issues using mine. So, the
2006 Apr 11
4
Why is the internet connection important to LAN and PSTN calls?
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before the Cisco
79[46]0's start to ring.
If we were lucky enough to get a call through, then we could not transfer
the call, or place the call on hold, or park the call.
Outbound calls seemed to have a delay
2006 Feb 14
4
BRI Newbie - What Hardware, PCI, in the US?
We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This
should get rid of static on the line (at least any static generated by our
half of the circuit), right?
I am a total virgin to ISDN. I understand that we need two BRI circuits to
provide four voice channels, and that the hardware to speak to the BRI
circuits can be passive or active, with the active type being much more
2006 Feb 16
2
Random Hangups/Disconnects
Well, I thought and hoped my issue of random hangups on our TDM400P were
related to busydetect=yes in zapata.conf. The behavior of a call being
hungup has not changed, however, since setting the busydetect option to
'no'. Again, the only affected user is my loud talker...
What are some causes/solutions to seemingly random call disconnects on Zap
channels that people have seen? I have
2006 Jan 03
2
Looping Problem With Call Forwards - Do you have comments on my solution?
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my extension at work, and my cell phone via NuFone.
Problem: A loop can be created if my cell phone is not on. Say a call comes
into my * box, it uses NuFone to call my
2010 Feb 17
1
One-Way Audio after Hold
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet,
and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet
parameters are all set correctly in sip.conf. An inbound call from Sipphone
works great until the local channel places the call on hold. During hold,
the Sipphone user cannot hear music, only silence. The silence continues
after the hold, though
2011 Mar 03
4
SIP Provider Recommendation in US
I am becoming frustrated with our current VOIP provider. Does anyone have
any suggestions for a provider that supports asterisk well and provides
solid service? Voip-info.org has a husge list of providers, but it is
impossible to tell the fly-by-night operations from the reputable providers.
--Brent
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2002 Nov 27
1
Upgrade failed? 2.2.5-pre1 to 2.2.7
Hello,
I downloaded the source, followed the instaructions to the T (make,
install, etc...). Everything seemed to go fine, I think the compilation and
installation were flawless (after all, I have compiled 2.2.5 sucessfully,
and havent done anything to this box as far as removing software libraries),
there were no reported errors. The install portion even had the messages
saying all went well,
2005 Sep 26
1
Socket 478 Motherboard for use with TDM400P
Well, it has been a long saga over here trying to get a TDM400P to work
with an Intel D845HV motherboard, and the towel is close to being thrown
in. Does anyone know of a (few) motherboards that work well with the
TDM400P and are socket 478?
--Brent
2006 Mar 23
1
RE: MeetMe freezes machine with Junghanns
Dollars to donuts it is related to these two posts, but no one seems to know
where or why it happens - this issue doesn't seem to be related to one
specific piece of hardware:
Post 1)
*********************************************************
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while