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T38 Fax Detection Using g729 and Asterisk VoIP Software

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 to the fax extension. For Asterisk to detect the Fax Tones you will need to set faxdetect to either yes or cng, you will also require using alaw or ulaw codec.

It might be suggested that you configure your incoming calls from Eutelia to go directly to the fax receive function whilst having the g729 codec enabled, I expect you will then see T.38 re-invite come from Asterisk. (more…)

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

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
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.0.2:5060:
INVITE sip:1234568@172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
Max-Forwards: 70
From: “asterisk” ;tag=as1689b2b6
To:
Contact:
Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Date: Sat, 14 Apr 2012 09:10:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1594270426 1594270426 IN IP4 172.16.0.1
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 172.16.0.1
t=0 0
m=audio 19486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Can someone tell me what is this issue ?

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 >>
>
> — Executing [00918885268942@default:2] Dial(“Console/dsp”, “SIP/
> 00918885268942@voipon”) in new stack
>
> == Using SIP RTP CoS mark 5
>
> Audio is at 10.30.131.136 port 12556
>
> Adding codec 0x2 (gsm) to SDP
>
> Adding codec 0x4 (ulaw) to SDP
>
> Adding codec 0x8 (alaw) to SDP
>
> Adding non-codec 0x1 (telephone-event) to SDP
>
> Reliably Transmitting (NAT) to 217.14.138.127:5065:
>
> INVITE sip:00918885268942@sip.voipon.co.uk:5065;user=phone SIP/2.0
>
> Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport
>
> Max-Forwards: 70
>
> From: “asterisk” ;tag=as2f61c90c
>
> To:
>
> Contact:
>
> Call-ID: 3cd12da658b42c10186c01ed3a7d21a7@sip.voipon.co.uk
>
> CSeq: 102 INVITE
>
> User-Agent: Asterisk PBX 1.6.2.21
>
> Date: Fri, 03 Feb 2012 06:01:16 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>
> Supported: replaces, timer
>
> Content-Type: application/sdp
>
> Content-Length: 313
>
>
>
> v=0
>
> o=root 1850926672 1850926672 IN IP4 10.30.131.136
>
> s=Asterisk PBX 1.6.2.21
>
> c=IN IP4 10.30.131.136
>
> t=0 0
>
> m=audio 12556 RTP/AVP 3 0 8 101
>
> a=rtpmap:3 GSM/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=silenceSupp:off – – – -
>
> a=ptime:20
>
> a=sendrecv
>
>
>
> —
>
> — Called 00918885268942@voipon
>
> Retransmitting #1 (NAT) to 217.14.138.154:5060:
>
> INVITE sip:00918885268942@sip.voipon.co.uk:5065;user=phone SIP/2.0
>
> Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport
>
> Max-Forwards: 70
>
> From: “asterisk”
;tag=as2f61c90c
>
> To:
>
> Contact:
>
> Call-ID: 3cd12da658b42c10186c01ed3a7d21a7@sip.voipon.co.uk
>
> CSeq: 102 INVITE
>
> User-Agent: Asterisk PBX 1.6.2.21
>
> Date: Fri, 03 Feb 2012 06:01:16 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>
> Supported: replaces, timer
>
> Content-Type: application/sdp
>
> Content-Length: 313
>
>
>
> Scheduling destruction of SIP dialog ‘
> 3cd12da658b42c10186c01ed3a7d21a7@sip.voipon.co.uk‘ in 32000 ms (Method:
> INVITE)
>
> — SIP/voipon-00000014 is circuit-busy
>
> Scheduling destruction of SIP dialog ‘
> 3cd12da658b42c10186c01ed3a7d21a7@sip.voipon.co.uk‘ in 32000 ms (Method:
> INVITE)
>
> == Everyone is busy/congested at this time (1:0/1/0)
>
> — Executing [00918885268942@default:3] NoOp(“Console/dsp”,
> “**CONGESTION**”) in new stack
>
>
> –
>
> Thanks and regards
>
> Virendra Bhati
> +91-8885268942
> Software Engineer
> E-mail-: virbhati@gmail.com
> Skype id:- virbhati2
>
>

Compile error 1.8.8.1

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
indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o
netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o
sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o
stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o
tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o
ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a
-> asterisk
pbx.o: In function `ast_str_substitute_variables_full':
/usr/src/asterisk/asterisk-1.8.8.1/main/pbx.c:3786: undefined reference to
`ast_str_expr’
collect2: ld returned 1 exit status
make[1]: *** [asterisk] Error 1
make: *** [main] Error 2

Without making any further changes, compiling 1.8.8.0 worked without any
errors. It seems, this is stemming from compiling:

[CC] xmldoc.c -> xmldoc.o

I ran with “–disable-xmldoc” flag while running “./configure”, and this
time also compilation failed with the same error. I’ve reverted back to
using 1.8.8.0.

Any suggestions on why this is happening, or what am I doing wrong?

Regards
HASSAN

Problems with codec translation when using Monitor and MixMonitor

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: No translator path from alaw to unknown
[Jan 11 17:55:48] WARNING[19500]: translate.c:256
ast_translator_build_path: No translator path from alaw to unknown

testpbx*CLI> g729 show licenses
0/1 encoders/decoders of 1 licensed channels are currently in use

Licenses Found:
File: G729-… — Key: G729-… — Host-ID: … — Channels: 1 (Expires:
…) (OK)

Logs with Monitor:
[Jan 11 17:49:49] WARNING[19491]: translate.c:256
ast_translator_build_path: No translator path from alaw to g723
[Jan 11 17:49:49] WARNING[19491]: file.c:186 ast_writestream: Unable to
translate to format wav49, source format g729

testpbx*CLI> g729 show licenses
0/1 encoders/decoders of 1 licensed channels are currently in use

Licenses Found:
File: G729-…lic — Key: G729-… — Host-ID: … — Channels: 1 (Expires:
…) (OK)

I’ve searched for this on forums but can’t find a complete answer yet.

Thanks!

Elder D. Arohuanca

Lima – Peru

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

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
callerid=”phone2<102>”
disallow=all
allow=ilbc
allow=g723
allow=gsm
allow=g723
allow=ulaw
allow=alaw
allow=adpcm
allow=h263
allow=h263p
allow=h261

In extension.conf

exten => 102,1,VoiceMail( 102@default,u )

exten => 102,n,Hangup()

exten => 704,1,VoiceMailMain()

In voicemail.conf

format=h263|alaw|gsm

101 =>1234,phone1
102 =>1234,phone2

I call to 102 and save video voice mail

check in 102/INBOX

    267 Dec 29 13:22 msg0020.txt
   61503 Dec 29 13:22 msg0020.h263

    330 Dec 29 13:22 msg0020.gsm
  1600 Dec 29 13:22 msg0020.alaw

My Xlite phone codecs

aLaw,uLaw,H.263

Thanks&Regards

Durgesh Mishra