What version of Asterisk. The "canreinvite" setting has changed in newer
versions. You need "directmedia=no" in newer versions to force asterisk to stay
in the path.
-----Original Message-----
From: nedi [[email protected]]
Received: Sunday, 29 Sep 2013, 6:17AM
To: Astlinux [[email protected]]
Subject: [Astlinux-users] one way voice problem if the transfer message enabled
and parallel calls to internal ext.
Hi,
I have problem with my pbx and Gigaset C610 IP Phone on internal number 71.
If the incoming call coming to the phone Number 41712345670 and the call ist
forwarded to internal numbers as parallel call included internal number 71
after 16 second there is the problem with only one way voice, the caller can’t
hear the person on the extensions 71. But If the call coming direct to this
extension 41712345671 there is no audio problem both person can hear each other.
I have testet this with a snom 300 Phone and there is no one way audio problem.
The Gigaset phone don't have gsm codes and on pbx i have disabled gsm codec
I have tested if I disable transfer message, after disable transfer message
there is no problem with one way voice
;exten =>70,n,Playback(transfer)
I have tried transfer message with ulaw, alaw and gsm codec all the same
problem i can hear the message but after transfer there is only one way audio.
has anyone ideaa what can I try to fix that?
In my SIP.Conf i have
[general]
useragent=MYPBX2SG
port=5060
context = from-sip-external; send unknown sip callers to this context
alwaysauthreject=yes
deny = 0.0.0.0/0.0.0.0
permit = 192.168.1.0/255.255.255.0
allowguest=no
language=de
disallow=all
allow=ulaw
allow=alaw
maxexpiry=120
defaultexpiry=50
qualify=2000
srvlookup=yes
nat=yes
allowsubscribe = yes
subscribecontext = hints
[41712345670]
type=peer
username=41712345670
secret=xxxxxx
fromuser=41712345670
host=siphost.ch
fromdomain=sipdomain.ch
canreinvite=no
insecure=port,invite
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=incominmygsipprovider
dtmfmode=info
[41712345671]
type=peer
username=41712345671
secret=xxxxxx
fromuser=41712345671
host=siphost.ch
fromdomain=sipdomain.ch
canreinvite=no
insecure=port,invite
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=incominmygsipprovider
dtmfmode=info
[70]
type=friend
username=70
secret=xxxxx
callerid="70" <70>
host=dynamic
mailbox=70@default
dtmfmode=info
canreinvite=no
insecure=port,invite
context=70
disallow=all
allow=ulaw
allow=alaw
callgroup=2
pickupgroup=2
notifyringing=yes
callcounter=yes
limitonpeers = yes
[71]
type=friend
username=71
secret=xxxxx
callerid="71" <71>
host=dynamic
mailbox=70@default
dtmfmode=info
canreinvite=no
insecure=port,invite
context=71
disallow=all
allow=ulaw
allow=alaw
callgroup=2
pickupgroup=2
notifyringing=yes
callcounter=yes
limitonpeers = yes
in my Extensions.conf I have
[incomingmysipprovider]
exten => 41712345670,1,Dial(local/70@70)
exten => 41712345671,1,Dial(local/71@71)
exten =>70,1,Dial(SIP/70,16,r)
exten =>70,n,Playback(transfer)
exten
=>70,n,Dial(SIP/71&SIP/72&SIP/73&SIP/74&SIP/75&SIP/76&SIP/77&SIP/78&SIP/79,13,r)
exten =>70,n,Playback(vm-nobodyavail)
exten =>70,n,Voicemail(70)
exten =>70,n,Hangup
exten =>71,1,Dial(SIP/71,18,r)
exten =>71,n,Playback(transfer)
exten =>71,n,Dial(SIP/70,13,r)
exten =>71,n,Answer
exten =>71,n,Playback(vm-nobodyavail)
exten =>71,n,Voicemail(71)
exten =>71,n,Hangup
best regards nedi
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------------------------------------------------------------------------------
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Free Intel webinars can help you accelerate application performance.
Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most from
the latest Intel processors and coprocessors. See abstracts and register >
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_______________________________________________
Astlinux-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
[email protected].
