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 ------------------------------------------------------------------------------ October Webinars: Code for Performance 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 > http://pubads.g.doubleclick.net/gampad/clk?id=60133471&iu=/4140/ostg.clktrk _______________________________________________ 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].
