Displaying 20 results from an estimated 10000 matches similar to: "Asked to transmit frame type 256, while native formats is 0x4"
2006 Mar 16
1
Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)
Hi everyone,
I have an issue which is kind of a catch 22 situation. I had outgoing
calls to my new PSTN provider working perfectly. Then I started
focussing on incoming calls. It seems that I can solve an error which
gets my incoming calls working but that in turns means my outgoing calls
don't work. - Strange.
Anyhow I was getting an error:
Process_sdp: No compatible codecs!
And from
2010 Nov 19
0
help with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)
Hi all,
i have a little problem to understand this warning message, it's annoying
and it cause a lot of spurious in the log files.
Im working with this scenario:
a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are
always routed to this.
a list of sip UAs that potentially can use any codec apart g729/g723.
I setup the asterisk to do as mediaproxy so directmedia=no and
2011 Mar 09
3
Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)
Hello!
Client is using ulaw, however server sometimes fills the log with following:
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit
frame type slin, while native formats is 0x8 (alaw) read/write = 0x4
(ulaw)/0x8 (alaw)
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit
frame type slin, while native formats is 0x8 (alaw) read/write = 0x4
(ulaw)/0x8 (alaw)
2003 Oct 30
0
SIP error: Asked to transmit frame type 64
Hi there,
I'll need some help with this: Trying to establish an IAX2 link between
two servers works in one direction (SIP client with ulaw), but not in the
other (SIP client with GSM). The client used for this is X-Lite behind
NAT while both servers have a public IP (I playback an anouncement before
trying to connect to the second *).
Error on the originating * server:
2011 Jun 28
1
Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Asterisk 1.8.3.2
I have been getting this warning constantly on CLI in a call scenario where
I use local channels to connect SIP with PSTN.
I use callfile and local channel to first call a PSTN number and if
answered, use local channel to call SIP phone with music on hold enabled in
Dial string.
If I call PSTN from SIP directly or vice versa I don't see this warning
coming.
On SIP I have
2006 May 26
0
SIP call problem
Hello,
I have problem to make SIP calls, i have asterisk +
PC InterP4 + Digium TDM400P
here is the content of the sip.conf:
[SIP_PROVIDER]
type=peer
fromuser=testcomclient
username=testcomclient
secret=testr
host=IP_SIP_PROVIDER
;dtmfmode=rfc2833
context=interne
canreinvite=no
;allerid=Beer
disallow=all
allow=ulaw
allow=gsm
allow=g723.1 ; Asterisk only
2015 Jul 15
2
Problem "no voice"
Hi list!
I have 4 numbers on my Asterisk 1.8.
3 work perfectly, the 4.th not.
I'm not sure, when it finish to work, since a month ago it runs without any
problem...
Well, if I will be called on this number I can't hear anything and in
Asterisk I see these:
[Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format
2005 Jun 08
2
format g729 and Voxee.com
Hi,
I have just signed up with Voxee.com and have attached my Asterisk
server to dial them via IAX2.
Below is the start of the log which dials the number and promply
hangs up when the call is answered, with the logs saying that the
channel is not compatiable.
I have traced this down to the g.729 codec which I don't have
installed. Any ideas on how to force that the codec not be used?
2005 Aug 10
1
Error while calling
Dear all,
I am getting the below errors when using asterisk. I am using sjphone for testing purpose.
Below are the setting for sip.conf and extension.conf
When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect.
Can anybody tell me what does this error means and the how to solve this issue.
Thanking You,
Joel
sip.conf
[general]
context=default
2010 Feb 19
1
transcoding with TC400P
Hello,
I have transcoding card TC400P installed in server running Debian with
Asterisk 1.4.23. Everything seams to be fine and after I boot up
server I see in dmesg:
7.590966] Zapata Telephony Interface Registered on major 196
[ 7.590966] Zaptel Version: 1.4.12.1
[ 7.590966] Zaptel Echo Canceller: MG2
[ 7.610963] zttranscode: Loaded.
[ 7.618969] wctc4xxp: tc400b0: Attached to
2005 Jul 13
1
Suddenly a problem with outgoing calls made from Cisco phones...
Hi all!
Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building...
The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and
after about a minute the phone
2005 Jul 11
0
error related to the native formats
Hello friends,
i make a call through queue to the agent
when agent lifts the call it gives one side voice and
i get this message in the debug
chan_sip.c:1880 sip_write: Asked to transmit frame
type 64, while native formats is 4 (read/write = 4/4)
in my sip.conf iam allowing only ulaw
can any body suggest where i
may be missing
with regards
RK
2006 Apr 19
0
sip.conf codecs: ulaw, alaw and g729
Hi,
When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw)
stop working and I get the frame type error for them, but g729 works fine.
I've cleared general part of sip.conf of codec info to be on safe side. If
ulaw and alaw are the only ones allowed they work fine. Asterisk shouldn't be
doing any encoding or decoding, all codecs should be passing through. Any
2006 Mar 14
1
Codec Issue
Hi,
I have an issue which is kind of a catch 22 situation. I had outgoing calls
to my new PSTN provider working perfectly. Then I started focussing on
incoming calls. It seems that I can solve an error which gets my incoming
calls working but that in turns means my outgoing calls don't work. -
Strange
Anyhow I was getting an error:
Process_sdp: No compatible codecs!
And from the SIP
2006 Mar 29
1
Oneway Audio
Hi all,
I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk
SVN-trunk-r15187 to avail the PARKEDAT variable.
- I park the call using ParkAndAnnounce
- plays moh.
- accept the call using ParkedCall
The following errors are coming on the console and there is oneway audio -
no audio after Music-On-Hold at caller's side. Please advice.
I am testing using cisco 7902
2010 Jul 20
3
Problem with SIP
Good afternoon list.
I'm experiencing a problem with my SIP channel's. When I have an external
connection for one of my SIP carrier's, I can listen to the client and the
client listens to me normally. The problem is when I will transfer this
connection, the call is mute for the extension I have transfered. Only the
client hears normally. In the console of Asterisk generates the
2011 Apr 21
1
Transcode ulaw/g722 problem
We are getting the following on 1.8.3 and 1.8.4-rc2, HELP!
Why is Asterisk unable to transcode to/from ulaw and g722?
[2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722)
[2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,
2010 Nov 12
1
Call failed becaus of SIP tanslate
Hi Guys,
I have a the following issue when I ma trying to place a call through my
voip provider, I am using an asterisk 1.2.21.1, do you have an idea what
could fix this issue (as you can see when the other party answered, the call
get dropped because of probably sip incompatibility)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats
2004 Jul 01
2
Grandstream HT286 1.0.4.63 & Meetme
Good day!
Have a weird problem with HT-286 and Conference room. I use Asterisk
CVS-HEAD-06/04/04.
Here it is:
When HT-286 get into the conference room first and nobody in that room
everything seems ok (with any codec HT286 allowed), but when HT-286 get
into conference room when somebody already there, have got different HT
behavior:
1. When HT use GSM codec => it connects to conference room,
2015 Feb 23
2
Question about Warning message
Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write)
We found that line in function "sip_write" inside "chan_sip.c".
In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared