Hi
It could be that the RTP sessions aren't completely setup when you get
connected to the destination.
Kind Regards
Claus Futtrup
- Original Message -
From: "Jens Vagelpohl" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion&q
Hi,
1) 0.0.0.0 just means listning on all interfaces and their ip adresses, not
a problem.
2) Do a set verbose 100 to see if you have any communication with the sip
phones or startup asterisk with asterisk -vvvggg
3) This is because a MPG3 file used for music on hold isn't support or that
th
call is between SIP to SIP there's no
problem at all (set to RFC 2833).
Kind Regards Claus
Futtrup
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Hi there,
switch off G711 alaw codec then it should ok
Kind regards
Claus
- Original Message -
From: "Altus Syman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Thursday, September 23, 2004 3:37 PM
Subject: [Asterisk-Users]
Hi Guys,Im having some problems with a Wildcard
TE410P card.. During a call I getsome strange messages and the voice drops
out:Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write:
Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24
16:40:17 DEBUG[1101416512]: ch
span 2
No message like this for span 1..
Kind Regards
Claus Futtrup
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Claus Futtrup
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- Original Message
Hi,
I have this strange problem I need some help with.. It appears that I have
harddisk noise captured by a Digium TE410P card (Same problem on 2 identical
machines..) The machines are two Compaq Proliant DL320 G3's...
Does anyone else have this problem..
Kind Regards
Claus Fu
Hi Guys,
Im having some problems with a Wildcard TE410P card.. During a call I get
some strange messages and the voice drops out:
Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Write
returned -1 (Resource temporarily unavailable) on channel 1
Aug 24 16:40:17 DEBUG[1101416512]: ch
channel => 1-15,17-31 channel => 32-46,48-62
Kind Regards Claus Futtrup
- Original Message -
From:
Angel
Diaz
To: [EMAIL PROTECTED]
Sent: Thursday, August 12, 2004 11:35
PM
Subject: [Asterisk-Users] Question about
TE405P
Hi all, Does somebody kn
Hi there,
H maybe this is the problem:
11: 3655278467 XT-PIC uhci_hcd, eth0, t1xxp
Kind Regards
Claus Futtrup
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Here you go.
loadzone = no
defaultzone = no
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
bchan=48-62
dchan=47
Both E100P are connected to PSTN.
Kind Regards
Claus Futtrup
- Original Message -
From: "Storer, Darren" <[EM
only seem to
affect span 2. Users have been complaining about being unable to make calls,
but Im not sure if this has anything to do with that..
Please help.
Kind Regards
Claus Futtrup
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Version
I would be very interested in this, please let me know if I can help
Claus Futtrup
Project Manager
"The box said 'Requires Windows 95, NT, or better,' so I installed Linux."
This message is for the designated recipient only and may contain privileged
or confidential infor
Well that works.. But lets say I wont to be able to
control incoming and outgoing limits on all channels. I have 3 phones
registered and phone 1 calls phone 2. With the example below phone 1 cannot make
anymore calls.. But phone 2 can (even though stíll talking with phone
1)
Phone 2 can also
Hi there,
I was wondering how I can use setgroup and checkgroup for perfom incoming
and outgoing limitation checks.
I've have some users that doesn't what to be able to recieve more than 1
call at a time, and I also want to limit a users outgoing call abilities.
Any help would be greatly apprecia
age -
From: "Philipp von Klitzing" <[EMAIL PROTECTED]>
To: "Claus Futtrup" <[EMAIL PROTECTED]>
Sent: Wednesday, June 23, 2004 4:08 PM
Subject: Re: [Asterisk-Users] Problem with incominglimit and outgoinglimit
> Use SetGroup() and GetGroupCount() and
Hi,
I seem to have a problem with chanisavail and the call limits on sip
phones(incoming and outgoing)
The problem seems to be that chanisavail when trying create to create
channels and hanging them up afterwards screw up the current usage limit on
the phones.
Example with chanisavail:
Phone
Hi Guys,
Same problem here with latest CVS.
-cf
Hi,
Just started to get this error after updating to the latest CVS. Asterisk
dies if it can't create a channel - not so good.
-- Executing SetCallerID("SIP/750-2550", "39660426") in new stack
-- Executing Dial("SIP/750-2550", "CAPI/39660
Hi!
X-Lite: Menu --> Advanced settings --> Audio --> Silence
set keep transmitting after silence to 1 or something like that
Cf
- Original Message -
From: "Philipp von Klitzing" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 11:24 AM
Subject: Re: [Asterisk-U
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