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

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

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

[2206]
type=friend
secret=*******
callerid=2206
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
qualify=yes
disallow = all
allow = ulaw,alaw,h263,g729,gsm,h264
videosupport=yes

*codec list of asterisk 1.6.2.11*

*haddock8-astrx*CLI> core show codecs*
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME DESC