on 11/02/2011 07:44 AM Sammy Govind wrote the following:
core show application meetme
Thanks!
(I am new to asterisk, and just learning, so forgive my dumb questions)
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2011/11/2 Christian Tardif christian.tar...@servinfo.ca
Hi,
I have a 1.6.2.6 fax installation with a FFA license which seems to be
installed correctly (in fax show stats, I see that I have 1 Digium G.711
licensed channel, and 1 Digium T.38 licensed channel).
When trying to call my
Turn off faxdetect on this peer.
2011/11/2 Christian Tardif christian.tar...@servinfo.ca
Hi,
I have a 1.6.2.6 fax installation with a FFA license which seems to be
installed correctly (in fax show stats, I see that I have 1 Digium G.711
licensed channel, and 1 Digium T.38 licensed
Which phone model do you use? 192.168.33.90 doesn't belong to
192.168.23.0/24, so the first thing that i will be doing is check routing
tables on the phone. Or for more help you could draw you network layout and
poste here.
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Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff on
the code it seems to me, that in version 1.8.7 there is an autoanswer in
application dial
I use Polycom 501 phones. I have two networks - 192.168.23.0/24 and
192.168.33.0/24. My Asterisk server and most of my phones are on the 23
net. I have the one phone on the 33 net for cross-net testing (works fine on
1.4.41).
From: asterisk-users-boun...@lists.digium.com
Hi,
I'm running an Asterisk server connected to a carrier over 2 E1 cards. From
time to time, the Called Number Party presented by the carrier changes a
bit (for some reason I don't know) and is prefixed with a byte string (e.g.
: 00 34 34 39 ), which furtherly prevents libpri from getting the
On 02/11/2011 05:04, Anton Kvashenkin wrote:
Turn off faxdetect on this peer.
2011/11/2 Christian Tardif christian.tar...@servinfo.ca
mailto:christian.tar...@servinfo.ca
Hi,
I have a 1.6.2.6 fax installation with a FFA license which seems
to be installed correctly (in fax show
Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff
on
the code it seems to me, that in version 1.8.7 there is an autoanswer
in
Will this affect 10.X or is it just a 1.8 path?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Mudgett
Sent: Wednesday, November 02, 2011 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial
Hello list,
can anyone tell me what the following means (found in messages log) :
/[Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can't
setup media stream for this call.
[Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio
session: Address already in use
You have set an insufficient range in rtp.conf. Asterisk uses 2 ports per
call, but allocates 4 for transferring, etc, so when you set up a range of
10001-10040 (for example) you are basically setting a range of 10 concurrent
calls. Check rtp.conf and make the end range larger by 8 or 12 or
Hello,
thank you for your answer...
Current range (rtp.conf) : 11500 - 11650
Current calls : 20 à 25
Is this not sufficient ??
Jonas.
On 11/02/2011 04:00 PM, Danny Nicholas wrote:
You have set an insufficient range in rtp.conf. Asterisk uses 2 ports
per call, but allocates 4 for
Number of wished concurrent calls times 4 = Number of ports you'll
need to setup in rtp.conf ;-)
regards,
Ruben
Am 02.11.2011 16:05, schrieb Jonas Kellens:
Hello,
thank you for your answer...
Current range (rtp.conf) : 11500 - 11650
Current calls : 20 à 25
Is this not sufficient ??
150/4 = 37.5. maybe your sip peer has a conflicting range?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, November 02, 2011 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On 11/02/2011 04:13 PM, Danny Nicholas wrote:
150/4 = 37.5. maybe your sip peer has a conflicting range?
Where do I set this range in my peer definition ? I don't think there is
such a parameter in sip.conf
To be perfectly clear, how many RTP-ports are needed in the below
situation :
As I understand it, the scenario you describe would only use 2 channels (I
dont think the RTP channel gets established until connection; I could be
wrong about this as Asterisk might pre-reserve the channels for early media,
etc.) do keep in mind however that although you mention 2 and 2 (1
Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key
while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a
diff
on the code it seems to me, that in version 1.8.7 there is an
autoanswer in
Hi Richard,
Am Mittwoch, den 02.11.2011, 09:26 -0500 schrieb Richard Mudgett:
Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff
Il 02/11/2011 15.06, Philippe Sultan ha scritto:
PRI Span: 1 [70 13 a1 00 34 34 39 39 30 30 32 30 33 36 31 35 38 39 34
32 35]
Yes, like you guessed the third bit (wich is part of the called number
i.e.) is a NUL... but Q.931 allows any IA5 (ISO 646) character so it's a
bug in libpri not in
Sorry for the top post.
A valid fax extension is an extension named 'fax' in the incoming context for
that peer. I.e:
exten = fax,1,Verbose(1,Incoming fax call detected)
exten = fax,n,ReceiveFax()
You would obviously have a much longer definition, this was just a quick
example from my phone.
Giovanni,
Thanks a lot for clearing this up. The 's' extension would match any
number, and I would not be able to retrieve the actually dialed number from
within the dialplan, unless I'm missing something.
I'll file a ticket to solve that issue.
Thanks again,
Philippe
On Wed, Nov 2, 2011 at
Hello,
What is the method for changing the country for indications (eg. ring,
busy, etc. tones) in a dialplan?
Thanks,
Elliot
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2011/11/2 giovanni.v i...@keybits.org
snip
so it's a bug in libpri not in your telco side.
PRI Span: 1 [70 13 a1 00 34 34 39 39 30 30 32 30 33 36 31 35 38 39 34 32
35]
1. As the above line comes from libpri, how can one be certain the telco
side didn't send the weird NUL byte ?
(I don't
On 02/11/2011 17.52, Philippe Sultan wrote:
The 's' extension would match any
number, and I would not be able to retrieve the actually dialed number
from within the dialplan, unless I'm missing something.
No, obviously you missed none... but I prefer to route an incoming call
to a generic
On 02/11/2011 18.45, Olivier wroteo:
1. As the above line comes from libpri, how can one be certain the telco
side didn't send the weird NUL byte ?
Assuming libpri debug doesn't mess nothing is quite sure the NUL come in
from the telco.
2. Anyway, as Q.931 allows any IA5 (ISO 646)
Issue filed : https://issues.asterisk.org/jira/browse/PRI-128
Philippe
On Wed, Nov 2, 2011 at 7:00 PM, giovanni.v i...@keybits.org wrote:
On 02/11/2011 17.52, Philippe Sultan wrote:
The 's' extension would match any
number, and I would not be able to retrieve the actually dialed number
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