Hi
I have an issue with my Asterisk installation.
I have a SIP account at OVH that I want to be able to pickup on any authenticated softphone (my desktop computer, my smartphone and my laptop). I installed Asterisk 1.8.10.1 on my dedicated Utunbu server (using apt-get), created 3 accounts for each device and created a call group.
Everything looks fine except I can’t hear my interlocutor from the landline (but he can hear me). When I call a device from another (ie 6001 to 6002), it works fine.
Asterisk is installed on a dedicated server, directly connected to the Internet. All firewall ports are opened so it’s not a NAT/firewall issue. It looks like an issue between my Asterisk server and OVH.
Here are my config files :
sip.conf
[code][general]
bindport=5060
bindaddr=XXX.XXX.XXX.XXX ; My server’s IP, authentication fails when not provided
srvlookup=yes
defaultexpiry=3600
registertimeout=30
registerattempts=0
allowguest=no
encryption=no
nat = no
directmedia=no ; New name for "canreinvite"
directrtpsetup=no
; SIP trunks
register => 033285XXXXXX:********@sip.ovh.fr
[lenwe-ovh]
encryption=no
type=peer
host=sip.ovh.fr
context=lenwe-incoming
language=fr
insecure=invite,port
fromdomain=sip.ovh.fr
fromuser=033285XXXXXX
username=033285XXXXXX
secret=********
qualify=yes
dtmfmode=auto
language=fr
video=no
disallow=all
allow=ulaw,alaw,g729[/code]
users.conf
[code][general]
fullname = New User
userbase = 6000
hasvoicemail = no
hassip = yes
hasiax = no
hasmanager = no
lenwe
type=friend
host=dynamic
dtmfmode=auto
fullname = Lenwe
context=lenwe-internal
secret=O1mwODnU
language=fr
video=no
mailbox=XXXXXXXX@XXX.com
disallow=all
allow=ulaw,alaw,g729
6001
username = lenwe-office
6002
username = lenwe-mobile
6003
username = lenwe-laptop[/code]
extensions.conf
[code][general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[lenwe-incoming]
; from external
exten => s,1,Dial(SIP/6001&SIP/6002&SIP/6003,20,t)
[lenwe-internal]
; to internal
exten => _6XXX,1,Dial(SIP/${EXTEN},20,t)
; to external
exten => _X.,1,Dial(SIP/lenwe-ovh/${EXTEN}) [/code]
Here is the debug trace when placing a call from Asterisk to a landline phone (then hanging up).
Call from 6001 to landline 0951XXXXXX
== Using SIP RTP CoS mark 5
-- Executing [0951XXXXXX@lenwe-internal:1] Dial("SIP/6001-00000000", "SIP/lenwe-ovh/0951XXXXXX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/lenwe-ovh/0951XXXXXX
-- SIP/lenwe-ovh-00000001 is ringing
-- SIP/lenwe-ovh-00000001 is making progress passing it to SIP/6001-00000000
-- SIP/lenwe-ovh-00000001 answered SIP/6001-00000000
-- Remotely bridging SIP/6001-00000000 and SIP/lenwe-ovh-00000001
== Spawn extension (lenwe-internal, 0951XXXXXX, 1) exited non-zero on 'SIP/6001-00000000'
Now, this is the trace when calling from a landline to my SIP line. The 6001 and 6002 clients are up and ringing (6003 is offline).
Call from landline 0951XXXXXX to 6XXX
== Using SIP RTP CoS mark 5
-- Executing [s@lenwe-incoming:1] Dial("SIP/lenwe-ovh-00000005", "SIP/6001&SIP/6002&SIP/6003,20,t") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6001
== Using SIP RTP CoS mark 5
-- Called SIP/6002
[Dec 12 12:56:13] WARNING[15946]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
-- SIP/6001-00000006 connected line has changed. Saving it until answer for SIP/lenwe-ovh-00000005
-- SIP/6002-00000007 connected line has changed. Saving it until answer for SIP/lenwe-ovh-00000005
-- SIP/6001-00000006 is ringing
-- SIP/6002-00000007 is ringing
-- SIP/6001-00000006 connected line has changed. Saving it until answer for SIP/lenwe-ovh-00000005
-- SIP/6001-00000006 answered SIP/lenwe-ovh-00000005
== Spawn extension (lenwe-incoming, s, 1) exited non-zero on 'SIP/lenwe-ovh-00000005'
Do you know how could I fix this issue ?
Thanks