I have a SIP trunk going from an Asterisk 11 installation to an ADTRAN 7100. Everything works if you don’t place someone on hold or transfer the call. The re-invites lose audio both directions. I also had this problem with version 1.6 so I upgraded to the latest version (11.0.0).
This is on a local network (10.132.0.0/16). No routers or NAT between them. The phones connected to the 7100 are connected to their downstream ports (1-24).
The upstream router port is 10.132.10.14.
The 7100 SIP media gateway is at 10.132.21.254/16
The Asterisk Linux Server is on 10.132.6.15
As I see the problem: The initial call is as expected. The placing on hold is also as expected, when the call is taken off of hold, the call flow all looks right, but asterisk never starts sending RTP on the invited and 200 OK’d IP/Port. Looking through the wireshark packets, there is an RTP flow, but it is still pointing at the stream established in the HOLD invite at 18.243 seconds. Asterisk had already received an invite and issued an OK for th enew stream, but never switched. Is there something in my SIP config that could be causing this? Why do the first and second invites work fine, but the third doesn’t?
Thanks,
Mike
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This is the SIP.conf entry:
[xxxxxx]
username=xxxxxx
secret=xxxxx
type=friend
port=5060
nat=never
context=xxxxx
insecure=port,invite
host=10.132.21.254
fromuser=xxxxx
;defaultexpirey=20
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
This is the call flow:
|Time | 10.132.6.15 | 10.132.10.14 |
| | | 10.132.21.254 |
|9.432 | INVITE SDP (g711A g711U telephone-eventRTPType…1) | |SIP From: “Mike Myhre” <sip:sss7100@10.132.6.15 To:<sip:2383@10.132.21.254:5060
| |(5060) ------------------> (5060) | |
|9.440 | 100 Trying| | |SIP Status
| |(5060) <------------------ (5060) | |
|9.696 | 180 Ringing | |SIP Status
| |(5060) <------------------ (5060) | |
|11.171 | 200 OK SDP (g711U telephone-eventRTPType-101) | |SIP Status
| |(5060) <------------------ (5060) | |
|11.171 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|11.454 | RTP (g711U) | |RTP Num packets:333 Duration:6.638s SSRC:0x72645C86
| |(10400) <-------------------------------------- (50598) |
|11.505 | RTP (g711U) | |RTP Num packets:334 Duration:6.717s SSRC:0x5427728E
| |(10400) --------------------------------------> (50598) |
|18.229 | INVITE SDP (g711U g729 telephone-eventRTPType-…) | |SIP Request
| |(5060) <------------------ (5060) | |
|18.229 | 100 Trying| | |SIP Status
| |(5060) ------------------> (5060) | |
|18.230 | 200 OK SDP (g711U telephone-eventRTPType-101) | |SIP Status
| |(5060) ------------------> (5060) | |
|18.243 | RTP (g711U) | |RTP Num packets:1075 Duration:21.477s SSRC:0x5427728E
| |(10400) ------------------> (10000) | |
|18.330 | 200 OK SDP (g711U telephone-eventRTPType-101) | |SIP Status
| |(5060) ------------------> (5060) | |
|18.388 | ACK | | |SIP Request
| |(5060) <------------------ (5060) | |
|18.401 | RTP (g711U) | |RTP Num packets:434 Duration:8.658s SSRC:0x1D861C07
| |(10400) <------------------ (10000) | |
|27.081 | INVITE SDP (g711U g729 telephone-eventRTPType-…) | |SIP Request
| |(5060) <------------------ (5060) | |
|27.081 | 100 Trying| | |SIP Status
| |(5060) ------------------> (5060) | |
|27.081 | 200 OK SDP (g711U telephone-eventRTPType-101) | |SIP Status
| |(5060) ------------------> (5060) | |
|27.181 | 200 OK SDP (g711U telephone-eventRTPType-101) | |SIP Status
| |(5060) ------------------> (5060) | |
|27.283 | ACK | | |SIP Request
| |(5060) <------------------ (5060) | |
|27.352 | RTP (g711U) | |RTP Num packets:666 Duration:13.298s SSRC:0x5BF105B1
| |(10400) <-------------------------------------- (50598) |
|40.598 | BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|40.603 | 200 OK | | |SIP Status
| |(5060) <------------------ (5060) | |