Displaying 20 results from an estimated 400 matches similar to: "Rtp WARNING Messages on the Cli in safe_asterisk"
2004 Jun 30
3
Support for CENTOS-3.1
Hi,
Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which is clone of Redhat Enterprise 3.1 server.?
--
Best regards,
Frankie (fgravato@cfsdigital.com)
mailto:nanog@cfsdigital.com
2004 Jan 24
2
Sipura 2000 Transmit Issues? No Sound being passed to caller
I've been beating my head for 5 hours to figure out why my asterisk
server or sipura isn't passing my voice over to the caller. It seems i
can hear the caller but they can't hear me it seems either the
asterisk or the sipura isn't passing this information.
Here's my setup specs
asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service -
Voicepulse
2004 Jan 26
0
Anyone run * on OS X ?
With the 1U Apple G5 server at a good reliability/storage/pricepoint
level... got to thinking about compiling Asterisk on OS X.. at least for SIP
phone call switching, voicemail, etc. Has anybody attempted this? Email me
off list if this is too dev-heavy for the user list.
Thanks,
Ted W
-----Original Message-----
From: asterisk-users-request@lists.digium.com
2004 Jan 29
3
good job on the list server!
Brian,
Great job fixing up the list server... postings are happening very
quickly now (within minutes)!
Thanks,
Rich
2004 Jan 18
2
Nufone not taking GSM CALLS
Is nufone having problems taking gsm calls today
i had some issues dialing overseas to call my folks.
here's snip of what the console displayed
-- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack
Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1
2004 Aug 19
1
Received packet with bad UDP checksum
I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that
time I heard several "pops", or "clicks". Each time it happened, I saw
the following message:
Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP:
Received packet with bad UDP checksum
Any ideas what causes these, and why they turn in to a "pop", instead of
just silence, or a
2010 May 28
2
Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue
Hi folks,
I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD
4.7 -release. Everything seems to work fine. I have a macro which
answers, receives the fax to a tiff, and then runs a script (mailfax) to
convert that to pdf and email it. It all works perfectly except for some
errors I am seeing in the console. After it hangs up I get a dozen or so
messages in the cli
2009 May 13
1
Asterisk 1.6 T.38 generation towards a Cisco voice router
Hello List.
We are having some problems using t.38 together with a Cisco voice router at one of our providers end.
We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through.
When we send faxes to our other provider, who has cisco hardware
2004 Jul 09
1
RE: ATA 186, firmware SIP 3.1 and codec g.726 + now SIPURA SPA-2000
To me it's a error if I can't complete calls using the ATA configured to use
the g726 codec.
I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I
received NOTICES and WARNINGS, but I can't complete a call.
On a zap channel:
-- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack
-- Called 1/2217008
-- Zap/1-1 answered
2011 Nov 21
1
video calls not working
Hi list,*
*I am not able to make video calls between two sip accounts. below is the
information. please help me where I am missing the configuration.*
Extensions.conf*
exten => 111,1,Answer()
same => n,Dial(SIP/2206,60,r)
same => n,Hangup()
*SIP.conf*
[2218]
type=friend
secret=*******
callerid="Virendra" <9172341457>
host=dynamic ;
2007 Jun 26
1
Asterisk to Cisco 2600 GW DTMF Not Working
Hi All,
I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
with a PRI card in it, handing off to a PBX and vise verse. Calls in
and out are working fine except for DTMF from Asterisk to the 2600.
DTMF from the 2600 to Asterisk is fine.
Here are the Asterisk console warnings I get when I send DTMF from
Asterisk to the 2600:
== Forcing Marker bit, because SSRC has changed
Jun
2007 Jun 27
0
Asterisk to Cisco 2600 GW DTMF Not Working, Working now
On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote:
> Hi All,
>
> I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
> with a PRI card in it, handing off to a PBX and vise verse. Calls in
> and out are working fine except for DTMF from Asterisk to the 2600.
> DTMF from the 2600 to Asterisk is fine.
>
> Here are the Asterisk console warnings
2004 Sep 13
4
Unknown RTP codec 72 received
Hi all,
I get "Unknown RTP codec 72 received" message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
over voicepulse connect (IAX) and to FWD echo test (SIP). But this
message only with one SIP client, others (X-Lite too) not giving this
message. All X-Lite settings are identical. Asterisk is last cvs version
This what I see in console
2004 Sep 27
0
Speex/ILBC buggy with * 1.0 and X-Lite/Pro?
I'm playing with codecs at the moment and have found some notices errors
when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was
getting garbled sound, but after changing magic number for both codecs
to 97 (as per
http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and
http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to
get normal voice. BUT,
2004 Jun 02
0
ast_rtp_read: Unknown RTP codec
Any one see these? Are they benign, or is some system tuning required
to remove them?
Can't seem to find a resolution in the archives. If you have a link, it
would be appreciated.
Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP
codec 19 received
Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP
codec 72 received
Jun 2 10:59:00 NOTICE[163044272]:
2004 Jan 14
2
Static Noise coming from Wildcard FXS: Wildcard TDM400P
I recently plugged in Phone to my TDM400P Card to test out something I
mostly use sip phones to interface with *. All of sudden I'm getting lot of
line static noise coming of the card is there any settings I should look at
or anything I need to do on the command line at this point I'm open to any
ideas
I'm running
0.7.1 on Redhat 9.0 machine.
Any insight would be greatly appreciated.
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following
2004 Jul 26
0
rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received
After just having update to the latest CVS I am getting the following
message when I call VoicemailMain():
-- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack
-- Playing 'vm-login' (language 'en')
-- Playing 'vm-password' (language 'en')
-- Playing 'vm-youhave' (language 'en')
-- Playing
2005 Jul 11
4
Video phone settings???
I have three video phones here for testing:
Extension 6003 is Eyebeam
Extension 6004 is a hard phone (model 8770)
Extension 6005 is a hard phone (model 8882)
Can anybody have a look at my settings and the output I get from all
kinds of dialings, please.
The sip settings for all phones is (user / password different):
[6003]
type=friend
username=6003
secret=pwd
qualify=200
nat=yes
host=dynamic
2014 Feb 03
1
Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, im using a Asterisk Server which is not behind NAT.
First i had problems with the fax detection. But this is now solved
after adding a wait(2) at the correct place. But i'm still unable to
receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too
short after the Fax session has started.
My sip.conf includes
[general]
allowguest=no
alwaysauthreject=yes
sendrpid=rpid