VOIP phone1 transfers the call to VOIP phone2. Audio From cell phone to VOIP phone2 works but not vice versa. Direct calls from any combination works. 'sip show channels" shows two active channels (alaw audio).
Also tried:
Cell phone->SIP provider->Asterisk->SIP provider->Cell phone2
All direct calls from phone A to phone B works. Within Asterisk and/or Asterisk <-> outside phone.
I try to be more specific:
Cell phone A calls to a Asterisk phone B. Audio works fine every time.
Phone B transfers call to a Phone C (this can be a phone within Asterisk or a Cell phone outside). Maybe 50% of times audio is missing from B to C.
This might be a NAT problem but a) why it happens occasionally and b) NAT is configured properly and it works for everything else.
This might be codec problem but a) all the channels have alaw codec which works. Even the transferred call.
Any peers on the same network as asterisk can have nat=no. The peer out to your sip provider should have nat=yes.
You probably don’t want to set directrtpsetup=yes, keep it at no.
You have a weird problem and I’m not sure where else to look to help you. If the media comes in, and stays going through asterisk (not being reinvited out), it should be able to hear wherever it ends up since * is staying in the media path and passing the stream through.
Are there any clues on the cli when this is happening?
If all peers are using the same codec (alaw as you stated), including your sip provider, it shouldn’t be a codec issue. You are setting disallow=all and allow=alaw so you know it is picking alaw for each call leg correct?
This is what sip show channels shows: (first and last queries are non-working calls and second and third are working calls. However I don’t see any difference).
XXX.X97.95.4 0942899750 588b8e12021 00104/00000 0x0 (nothing) No
192.168.1.34 6006 51c9a7d2375 00102/00000 0x8 (alaw) No Tx: ACK
192.168.1.32 6005 132613f23ae 00104/00103 0x0 (nothing) Yes Rx: BYE Done
XXX.X97.95.4 0942899750 68E9E4809E2 00101/00002 0x8 (alaw) No Rx: ACK
4 active SIP channels
.32 is operator’s phone which transfers the call, .34 is the destination phone. During the test, all voip phones are within the same subnet as Asterisk.
Cli does not show any error messages if that’s what you meant.
50/50 ratio is not exactly right. 4/5 transferred calls works. There is something I don’t understand (Firewall/router perhaps).
I’m aware of NAT parameter. These phones are also used outside LAN so NAT should be enabled. However during the tests LAN has been used to minimize possible problems.
All destination phones are cell phones (Nokia).
The last remark is not essential. The line stays for few secs and disappears automatically. It is just about the timing I have captured ‘sip show channel’ information.
Is there a way to increase max delay between # and 1 (* and 2) for feature set functions. By default max delay is rather small and fails cos lazy fingers
Sure I can use just # but…