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.

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…)

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 0×2 (gsm) to SDP
Adding codec 0×4 (ulaw) to SDP
Adding codec 0×8 (alaw) to SDP
Adding non-codec 0×1 (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

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 0×2 (gsm) to SDP
>
> Adding codec 0×4 (ulaw) to SDP
>
> Adding codec 0×8 (alaw) to SDP
>
> Adding non-codec 0×1 (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
>
>

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

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