T38 Fax Detection Using g729 and Asterisk VoIP Software

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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…

Asterisk Tips 2.7 years ago 0 Answers

Can't make Asterisk send authentication to remote peer on INVITE

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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]
type=friend
secret=abcde
host=dynamic
context=demo on 172.16.0.1 : [natty]
type=peer
host=172.16.0.2
fromuser=test
secret=abcde originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at 172.16.0.1 port 19486

Asterisk Users 3.4 years ago 2 Answers

Can someone tell me what is this issue ?

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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 wrote: > Call is not routing from server to destination.
>
>
> app8*CLI> console dial 00918885268942
>
> [Feb 3 06:01:15] NOTICE[28124]: 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 >>
>

Asterisk Users 3.6 years ago 0 Answers

Compile error 1.8.8.1

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Hi, While compiling 1.8.8.1, 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

Asterisk Users 3.7 years ago 2 Answers

Problems with codec translation when using Monitor and MixMonitor

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Hi folks, 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 1.8.7.1 Logs with MixMonitor:
[Jan 11 17:55:48] WARNING[19500]: translate.c:256
ast_translator_build_path:…

Asterisk Users 3.7 years ago 3 Answers

Help_In Voicemail , vedio play but voice is not here out.

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Hi all, 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 In Sip.conf videosupport=yes [phone1]
type=friend
host=dynamic
context= employees
mailbox=101@default
callerid="phone1<101>"
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
allow=alaw
allow=adpcm
allow=h263p
allow=h264
allow=h263 [phone2]
 type=friend
host=dynamic
context= employees
mailbox=102@default

Asterisk Users 3.7 years ago 0 Answers

video calls not working

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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 ; 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
dtmfmode=inband
insecure=invite,port
;context=outbound
context=bhati-test
qualify=yes
accountcode=123654789
disallow = all
allow = ulaw,alaw,h263,g729,gsm,h264
videosupport=yes…

Asterisk Users 3.8 years ago 4 Answers

Forcing a CODEC

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Hi folks, 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…

Asterisk Users 3.9 years ago 4 Answers

dynamically alter list of offered codecs (for faxing)

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Hello everybody! 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…

Asterisk Users 4.1 years ago 0 Answers

Question about codec re-negotiation in asterisk 1.4.X

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Hi all,
I have a major issue with a codec renegotiation in an asterisk 1.4.33.1
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…

Asterisk Users 4.2 years ago 1 Answer