I would not turn on nat? E.g. nat=no in general and turn on for individual 
extensions if necessary. 

Regards
Michael Knill

Sent from my iPhone

On 29/09/2013, at 9:16 PM, nedi <[email protected]> wrote:

> 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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