Using T.38 termination with Eutelia provider could present a scenario where you can receive faxes using g729, but not being able to receive it, even if you are able to receive it while using alaw. This makes you to start suspecting of your VoIP Software. It might be the case that the remote endpoint, Eutelia, will need to detect the Fax Tones and send the T.38 ReINVITE to you, they may not have T.38 enabled. If that's the case, your Asterisk installation will need to detect the Fax Tones so as to make the decision the incoming call is a fax and then switch…
This is a really simple problem that I just can't get to work. There
are two Asterisk servers with the following sip user and peer. When a
call is attempted, Asterisk is not sending authentication details in
response to the 401. Note, if the secret is blank on 172.16.0.2 test,
the INVITE succeeds. on 172.16.0.2: [test]
context=demo on 172.16.0.1 : [natty]
secret=abcde originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at 172.16.0.1 port 19486
Your Server Voipon isn't responding- See if internet is working fine, or
your Voipon sip trunk/peer is registered OK? On Fri, Feb 3, 2012 at 5:53 PM, virendra bhati
> app8*CLI> console dial 00918885268942
> [Feb 3 06:01:15] NOTICE: console_video.c:133 console_video_start:
> voice only, console video support not present
> -- Executing [00918885268942@default:1] Answer("Console/dsp", "") in
> new stack
> < < Console call has been answered >>
While compiling 220.127.116.11, I met the following error:
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o
btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o
mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o
recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o
recno/rec_utils.o -> libdb1.a
[LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o
asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o
bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o
config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o
enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o
fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o
I'm having problems when I try to record my calls using MixMonitor or
Monitor. Calls are working well and audio quality is good.
But I just can't get recorded audio in one leg with both applications. It
happens with internal calls too. As it seems, the problem is my g729
licensing escheme. (just one license installed) What is the least number of licenses that are needed per recorded call?
or what can I do to fix it? My asterisk version: Asterisk 18.104.22.168 Logs with MixMonitor:
[Jan 11 17:55:48] WARNING: translate.c:256
I am using to Xlite to save video voice mail.
when i retreive it, then only video show , no voice is here out.
Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box.
I did following configuration
*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*
host=dynamic ; This device needs to register
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow = all
allow = ulaw,alaw,h263,g729,gsm,h264
How can I take advantage of a high-bandwidth CODEC, like G.722, for
internal communications at my site, but use G.711 (alaw/ulaw) for all
other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX
trunks, I was hoping that it would be possible to specify alaw and
ulaw as the first two CODEC choices for the SIP phones, as well as in
their sip.conf configurations, but that I could use the IAX trunks
(with bandwidth=high) to force the phones to…
Lately I've had experiences that I'd like to share with you:
I did a some faxing over VOIP during the last two years. Not that much,
lets say 1 fax per day on average. The setup is Old analog fax machine < -> Linksys PAP2 ATA < -> Asterisk 1.2 < -> DSL < ->
VoIP Provider .... I would estimate that 80% of the faxes went through on the first try.
The rest aborted transmission with some communications error and needed
a second (or rarely a 3rd) try.
Then suddenly, faxing didn't…
I have a major issue with a codec renegotiation in an asterisk 22.214.171.124
setup, which leads me to ask a general question about asterisk 1.4.X codec
negotiation: asterisk can support a re-negotiation of a codec "on the fly"
through a re-Invite? If my SIP provider sends me a re-invite changing codec
from g729 to g711, asterisk properly handle the matter?
I see in the trace that asterisk responds 200 OK to the provider, but *does
not send the re-invite to the UAC, and stops to send rtp to the UAC*. In…