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

I have setup a trunk on freepbx and the outbound route. Everytime I
dial via the trunk, I get “all circuits are busy now”. Incoming calls
are working fine on the trunk.

This is my dial

9|XXXXXXX.

and these are my peer details

allow=ulaw&alaw
canredirect=no
disallow=all
dtmfmode=rfc2833
host=192.168.9.251
insecure=very
type=peer

Below are the logs. my trunk name is called fxosip

Using SIP RTP TOS bits 184

[0K == Using SIP RTP CoS mark 5

[0K -- Called fxosip/0799490994

[0K -- SIP/fxosip-00000015 is circuit-busy

[0K == Everyone is busy/congested at this time (1:0/1/0)

[0K -- Executing
[s@macro-dialout-trunk:20][1;36mNoOp("SIP/3000-00000014",
"Dial failed for some reason with DIALSTATUS = CONGESTION and
HANGUPCAUSE = 1") in new stack

 -- Executing [s@macro-dialout-trunk:21]
Goto("SIP/3000-00000014",
"s-CONGESTION,1") in new stack

 -- Goto (macro-dialout-trunk,s-CONGESTION,1)

 -- Executing [s-CONGESTION@macro-dialout-trunk:1]
Set("SIP/3000-00000014", "RC=1") in
new stack

 -- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto("SIP/3000-00000014", "1,1") in
new stack

 -- Goto (macro-dialout-trunk,1,1)

 -- Executing [1@macro-dialout-trunk:1]
Goto("SIP/3000-00000014",
"continue,1") in new stack

 -- Goto (macro-dialout-trunk,continue,1)

 -- Executing [continue@macro-dialout-trunk:1]
GotoIf("SIP/3000-00000014",
"1?noreport") in new stack

 -- Goto (macro-dialout-trunk,continue,3)

>
 -- Executing [continue@macro-dialout-trunk:3]
NoOp("SIP/3000-00000014", "TRUNK Dial
failed due to CONGESTION HANGUPCAUSE: 1 - failing through to other
trunks") in new stack

 -- Executing [continue@macro-dialout-trunk:4]
Set("SIP/3000-00000014",
"CALLERID(number)=3000") in new stack

 -- Executing [90722490994@from-internal:5]
Macro("SIP/3000-00000014",
"outisbusy,") in new stack

 -- Executing [s@macro-outisbusy:1]
Progress("SIP/3000-00000014", "") in
new stack

 -- Executing [s@macro-outisbusy:2]
GotoIf("SIP/3000-00000014",
"0?emergency,1") in new stack

 -- Executing [s@macro-outisbusy:3]
GotoIf("SIP/3000-00000014",
"0?intracompany,1") in new stack

[0K -- Executing [s@macro-outisbusy:4]
Playback(“SIP/3000-00000014”,
“all-circuits-busy-now&pls-try-call-later, noanswer”) in
new stack

 — Playing ‘all-circuits-busy-now.gsm’
(language ‘en’)

 == Spawn extension (macro-outisbusy, s, 4) exited non-zero on
‘SIP/3000-00000014′ in macro ‘outisbusy’
== Spawn extension (from-internal, 90722490994, 5) exited non-zero
on ‘SIP/3000-00000014′

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