sip sans retour.

[davlefou]

bonjour,
j'ai la configuration suivante. Le dialplan lance bien la requête sur cet utilisateur mais il ne trouve pas la ligne.
Ma configuration marche bien ne iax mais pas en sip, je ne comprend pas pourquoi.
Les appels partent bien.




[sab]
username=sab
secret=passwd
type=friend
host=dynamic
mailbox=sab@domaine.com
context=direction
callerid=mon noms <100>
;permit=0.0.0.0/0.0.0.0
;permit=192.168.0.0/255.255.0.0
requirecalltoken=auto
disallow=all
allow=g729
allow=ulaw
allow=alaw
;allow=gsm

[davlefou]

j'ai ce message dans la console: connected line has changed

[jean]

tu peux mettre plus de détail de la cli - c'est bizarre !
aussi, ca sent la farce protocolaire... je regarderai bien avec un ngrep sur l ip du client

[davlefou]

le dialpan s'auto, depuis une ligne iax2 vers la sip:



Connected to Asterisk 1.8.8.0~rc4-1digium0+1~lucid currently running on Sabrine (pid = 1329)
Verbosity was 0 and is now 14
-- Registered SIP 'david' at 192.168.0.170:5060
> Saved useragent "SFLphone" for peer david
-- Accepting AUTHENTICATED call from 41.225.249.147:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (ulaw|alaw|gsm),
> priority = mine
-- Executing [101@direction:1] Dial("IAX2/sabi-782", "SIP/david&SIP/sabi") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/david
== Using SIP RTP CoS mark 5
-- Called SIP/sabi
-- SIP/david-00000006 connected line has changed. Saving it until answer for IAX2/sabi-782
-- SIP/sabi-00000007 connected line has changed. Saving it until answer for IAX2/sabi-782




réciprocité de la sip vers la iax2 :



> doing dnsmgr_lookup for 'sip.ovh.fr'
-- Registered SIP 'david' at 192.168.0.155:5060
> Saved useragent "Cisco/SPA112-1.3.1(003)" for peer david
== Using SIP RTP CoS mark 5
-- Executing [201@direction:1] Dial("SIP/david-0000000c", "IAX2/sabi&IAX2/iaxAMDBizerte/201&IAX2/iaxVillasBizerte/201&IAX2/iaxXeonBizerte/201,tT") in new stack
-- Called IAX2/sabi
[2013-07-30 19:38:11] WARNING[28492]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
[2013-07-30 19:38:11] WARNING[28492]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
-- Called IAX2/iaxXeonBizerte/201
[2013-07-30 19:38:11] WARNING[28492]: app_dial.c:2385 dial_exec_full: Invalid timeout specified: 'tT'. Setting timeout to infinite
-- Call accepted by 41.225.249.147 (format gsm)
-- Format for call is gsm
-- IAX2/sabi-4141 is ringing
-- Call accepted by 41.225.221.129 (format g729)
-- Format for call is g729
[2013-07-30 19:38:11] NOTICE[1461]: chan_iax2.c:10923 socket_process: Rejected connect attempt from 41.225.221.129, request '{EXTEN}@ovh-iax' does not exist
-- Hungup 'IAX2/iaxXeonBizerte-16686'
-- Hungup 'IAX2/sabi-4141'
== Spawn extension (direction, 201, 1) exited non-zero on 'SIP/david-0000000c'



cela sonne bien, j'ai juste pas décroché.