insight. The following problem that I found using Asterisk appears to be improper
handling of a call put on hold when there is no music on hold:
[FXO gateway] [Asterisk] [IP phone]
|-------[INVITE s/SDP]---------------->|-------[INVITE s/SDP]---------------->| | | | |<--------[180 Ringing]----------------|<--------[180 Ringing]----------------| | | | |<----[183 Session Progress]-----------|<-----------[200 OK/SDP]--------------| | | | |<--------[200 OK/SDP]-----------------|------------[ACK]-------------------->| | |<=========== RTP ====================>| |------------[ACK]-------------------->| | |<=========== RTP ====================>| |
{IP phone puts caller on hold}
| |<-----[INVITE/held SDP]---------------| | | | | |-----------[200 OK/SDP]-------------->| | | | | |<------------[ACK]--------------------| |============ RTP (one-way)===========>| | | | | |----------[BYE]---------------------->| | | | | |<------------[200 OK]-----------------| |
When the IP phone puts the gateway on hold, Asterisk gets the re-INVITE with held media but Asterisk doesn't re-INVITE the gateway. The RTP traffic to the gateway stops so the gateway handles the condition as a lost connection. Shouldn't asterisk be forwarding the re-INVITE to the gateway unless MOH is enabled?
--- Begin Message --- I'm having some problems with a SIP FXO gateway working with Asterisk when a
call that involves the gateway is put on hold. This gateway was working up to a firmware
upgrade but I believe it may have been working for the "wrong reasons". Here is what
happens:
1. User calls in from PSTN to SIP FXO gateway.
2. FXO gateway calls extension on Asterisk.
3. SIP extension is rung and answered. (Canreinvite=no so voicepath goes
from SIP extension - to - Asterisk - to - SIP FXO gateway.)
4. SIP extension puts incoming caller on hold (SIP extension re-invites Asterisk with
SDP c=IN IP4 0.0.0.0)
5. Asterisk does not re-INVITE SIP gateway and apparently stops sending RTP
packets to the SIP gateway. (I'm not using music on hold)
6. Gateway thinks something is wrong because the RTP packets stop arriving and
sends a BYE to Asterisk.
7. PSTN caller is mad because they just got dropped. :)
I think that the pre-release firmware for my SIP FXO gateway was not doing the RTP
check so I was getting away with the above scenario without problems. The question is:
How can I get Asterisk to either re-invite the FXO gateway with "held media" or send
"silence" RTP packets?
Any ideas would be appreciated.
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