Displaying 20 results from an estimated 900 matches similar to: ""Inband DTMF is not supported on codec G.711 u-law. Use RFC2833""
2004 Nov 03
5
FireFly Problems
How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and
Info. RFC2833 is the default, so I left it that way. I also unchecked all
the codecs except g711ulaw to force that codecs usage. However, when I go to
place a call, I get this:
Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF
is not supported on codec G.711 u-law. Use RFC2833
Nov 3 13:18:44
2004 Jul 12
3
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
Hi can anyone help me on this error msg??
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
thnx
St
2003 May 07
2
MGCP broken
hi all
I'm being spammed by these messages in the console (see below) and sound
doesn't work with today's cvs. I rolled back a week, and it works fine. In
addition to the sound problems, I had to enable inband dtmf squelch on the
dilnk mgcp phones. if not, each pressed key was counted twice
NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP
ast_dsp_process
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2003 Aug 07
1
Warning Messages
hi,
i have connected a SNOM 200 to the asterisk. here are my settings,
Codecs
-------
Default codec - g.711u/g.711a
Packet size - 20ms
Negotiation - Interoperable
Type - 160
DTMF
----
Inband - negotiate
Outband - negotiate
Payload Type - 101
when a call comes to the SNOM or when making an outdial, following warning
messages are coming on asteisk,
WARNING[1209214400]: File dsp.c, Line 1198
2004 Aug 18
1
Hangups - SIGFPE in dsp.c
Hi,
I'm running the latest CVS HEAD version of asterisk, and I'm experiencing
hangups during voice conversation. This happens quite regularely and
often.
The problem is in dsp.c, line 1235, where it says
accum /= len;
But `len', at this point, is 0, resulting in a SIGFPE. The routine
ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is
setting p->fr.datalen to
2012 Nov 12
1
Can I make asterisk do inband and rfc2833 at the same time?
I know I wouldn't normally want this due to double tones, but my
upstream provider has an issue where they negotiate rfc2833 but then
send dtmf inband. I don't expect to get both at the same time, so is
there a way to make asterisk turn on both inband or rfc2833? Auto
doesn't work because it sees the rfc2833 in SDP then ignores inband for
the remainder of the call.
Thanks.
2006 Jan 19
1
DTMF Simultaneous Inband and RFC2833 performedby Asterisk => Duplicate tones
> I have seen the following effect in Asterisk, though: where
> it converts
> an inband DTMF (eg coming off a Zap channel) into an
> indication, it mutes
> the audio where that tone is. But sometimes it leaves a
> teeny bit of the
> tone behind.
>
> If you take such a call over say IAX to somewhere and then
> back out a Zap
> channel, you end up with the
2006 Jan 18
1
DTMF Simultaneous Inband and RFC2833 performed by Asterisk => Duplicate tones
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: Max Glucksmann (Fax del trabajo).vcf
Type: text/x-vcard
Size: 617 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060118/b1b5e895/MaxGlucksmannFaxdeltrabajo.vcf
2005 Jul 14
2
CVS HEAD voicemailbox full error
Anyone else has problems with CVS HEAD's from today with voicemail
hanging up silently without any debug/error messages when checked?
It also keeps insisting that the user's voice mailbox is full and can't
store more messages even if I clear/rebuild the
/var/spool/asterisk/voicemail stuff.
I've tried falling back to voicemail.conf entries from realtime
voicemail with the same
2006 May 26
3
using a billing system
Hello to all,
Im trying to use DeadAGI to implement billing with Asterisk2Billing.
Before the billing, I had something like:
exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider)
Now, with Asterisk2Billing would be something like this?
exten => _2XXXXXXXX,1,Answer
exten => _2XXXXXXXX,2,Wait,2
exten => _2XXXXXXXX,3,DeadAGI,a2billing.php
exten => _2XXXXXXXX,4,Wait,2
exten =>
2003 May 19
1
G.729 warning
hi !
I have asterisk with Licensed G.729 codec enabled. Whenever I make a
call using this codec a warning apears as,
WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect
process 256 frames
WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect
process 256 frames
WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect
process 256 frames
2003 Dec 01
1
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
What does it mean ??
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2005 Jul 06
3
OT: Congrats, Europe!
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
-------------- next part --------------
A non-text attachment was scrubbed...
Name: vahan.vcf
Type: text/x-vcard
Size: 287 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/357c6cce/vahan.vcf
2004 Sep 30
1
sipfriends in MySQL question/request
Greetings,
Is there a way to tie a specific sip username to a IP address when
authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1
USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile)
The reason is that I'm using Wellgate FXSes that have
second/third/fourth FXS ports bugged when I use a password, but work ok
when there is no password. Linking the username to a specific ip could
2004 Oct 06
2
Working Wellgate *SIP* 38xx/35xx hardware anyone?
I'm loosing hair at cosmic speed now for the past 10 days.
Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have
very buggy firmware possibly due to hastely done porting from H.323
firmware.
Is there anyone on this mailing list who was able to:
1. setup a 35xxA FXS with all ports authenticating properly with *?
or
2. setup a 38xx FXO to work as dial-in from pstn to
2004 Nov 23
4
Quick Questions - IVR=Auto Attendant?
Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press
1 for" thing. Sales is asking me how to word it and I've always used both
terms interchangeably.
2003 May 18
3
SNOM100 GSM again
OK I did some researches and tests with it, and finally:
I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console:
WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames
My conclusion is that the snom100 utilizes MSGSM
2012 Jan 13
1
LSI/3ware 9750-4i and multipath I/O
Hi,
I was wondering if anyone has successfully configured two lsi/3ware 9750-4i series controllers for multipathing under CentOS 5.7 x86_64?
I've tried some basic setups with both multibus and failover settings, and had repeatable filesystem corruption over a iscsi(tgtd) or nfs3 connection.
Any ideas?
Vahan
2004 Oct 01
1
BUG? no output from 'sip show users|inuse|active|subscriptions' when using MySQL auth
I'm authenticating against sipfriends in MySQL, and have just noticed
that none of the below commands return any output:
sip show users
sip show inuse
sip show active
sip show subscriptions
Is this a bug or something wrong on my side?
I'm using the stable 1.0 cvs
Vahan