I tried it but it doesn't help. Tx for the suggestion.
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Browse All TopicsWhen I type show channels there are 0 channels in use yet sip show channels returns the records below. Kindly explain. How do I clear them without a restart of Asterisk? Are they a problem?
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Try SIP SHOW PEERS
What you are looking at is not stale... it's the channel id for each registration.
http://www.voip-info.org/w
That is a good question. I just tried "sip show inuse" on my local asterisk box and, purely from experimentation, got these results:
1) SIP phone 4003 is calling 4004 (during ringing)
* Peer name In use Limit
4004 1/1 4
4003 1/0 4
2) After the call is answered (4003 is talking to 4004)
* Peer name In use Limit
4004 1/0 4
4003 1/0 4
I also tried googling for more information about "sip show inuse", but couldn't find anything meaningful.
By the way, I would hazard a guess that the problems you are seeing on your sip calls are due to incorrect routing of packets or firewall/NAT problems. It is just a guess, but your list for "sip show channels" has too many INVITE and BYE's shown in the "Last message" column. On my Asterisk servers most of the "Last message" entries are ACK's or OK's. If your routing or firewall policies are wrong, it is possible for one end point to send a SIP message, but the other end point to then not be able to respond to it - at least it tries to send a response, but the response is not received. Usually, when this is happening you will see the first end point re-send the request several more times. For INVITE's and BYE's there is always an ACK or an OK at the end of the dialogue so that is what you are most likely to see in the "Last message" column.
Thanks this is most informative.
The iptables looks like this - does it look inorder?
-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m tcp -p tcp --dport 5060:5082 -j ACCEPT
-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m udp -p udp --dport 5060:5082 -j ACCEPT
#-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m tcp -p tcp --dport 8000:20000 -j ACCEPT
-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m udp -p udp --dport 8000:20000 -j ACCEPT
-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m udp -p udp --dport 4569 -j ACCEPT
The rules look ok as far as I can tell. It may not just be firewall rules. It may depend on the IP addresses embedded within the SIP packets (The Contact header and Via's especially) and also on your network routing.
I think you would have to capture some SIP packets to confirm if the problem is like I suggested. Especially look for the tell-tale resending of the same request. I use the built-in packet logging mechanism in Asterisk. CLI commands: "sip debug" (v1.2) or "sip set debug" (v1.4). Type "help sip" for details of all the debug commands.
A diagram would help.
Is NAT involved? If it is, then check the advice on my web page here:
http://www.smartvox.co.uk/
ICMP ping is very unlikely to be relevant to SIP.
I like you web site tx for the link.
Nat's not involved.
VSP<-->Asterisk Server<-->Clients Asterisk Server
All 3 are not behind a NAT.
Any ideas please?
By the way when I try and send to an alternate VSP I get the error
frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end
I have confirmed that "Silence supresion" is off on their side. They say other * servers send to them fine. Their devices are Quintums.
Perhaps this can give a clue.
I don't get this error to any other VSP I send traffic too.
Can you capture a SIP packet and post it back here as an attached snippet. I just need to see one example of a packet that is getting sent multiple times - probably an INVITE or a BYE. In particular, I want to see what is in these three headers (they contain IP addresses/port used for returning SIP messages or responses):
Via
Contact
Record-Route
http://www.smartvox.co.uk/
Business Accounts
Answer for Membership
by: feptiasPosted on 2008-12-08 at 01:28:41ID: 23119347
This is a guess, but I wonder if these CLI commands might be relevant:
sip history
sip history off
I have never used the commands and the wiki seems to have no info about them, but I would guess that they somehow cause Asterisk to retain information about old SIP calls. Perhaps you turned on "sip history" without realising what it would do. This would make sense because some of the calls in your list show that BYE was the last message, so it looks like Asterisk is aware that they had finished.