Posted: Fri May 22, 2009 4:42 pm Post subject: [asterisk-users] No response to our critical packet problem
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect calls to internal office extensions (which still
go through asterisk) OR voicemail
2) The other 20+ phones in the same office on the same network have 0 problems.
Here's a SIP trace of the problem.
yyy.yyy.yyy.yyy is the outside NAT IP
xxx.xxx.xxx.xxx is the IP of my PBX
dddddddddd is the dialed phone number
sssssssssss is the source phone number
The peculiar thing is that asterisk sends an OK in response to an INVITE,
then the phone sends back an ACK, which asterisk seems to ignore
because it retransmits the OK message again
Then eventually the phone gives up and sends a BYE message.
Posted: Fri May 22, 2009 5:19 pm Post subject: [asterisk-users] No response to our critical packet problem
James Lamanna wrote:
Quote:
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect calls to internal office extensions (which still
go through asterisk) OR voicemail
2) The other 20+ phones in the same office on the same network have 0 problems.
Here's a SIP trace of the problem.
yyy.yyy.yyy.yyy is the outside NAT IP
xxx.xxx.xxx.xxx is the IP of my PBX
dddddddddd is the dialed phone number
sssssssssss is the source phone number
The peculiar thing is that asterisk sends an OK in response to an INVITE,
then the phone sends back an ACK, which asterisk seems to ignore
because it retransmits the OK message again
Then eventually the phone gives up and sends a BYE message.
-- James
I think I know what the problem is here. It's not the fault of the phone, but of
Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#',
specifically) to Asterisk. Asterisk only keeps track of the last incoming Cseq
in a dialog, so once the INFO arrives, we no longer have any memory of the Cseq
of the INVITE that the phone sent.
Later, we send a 200 OK response for the INVITE. Then, when we receive the ACK
from the phone, we drop it since it's Cseq is less than the latest Cseq we
received in this dialog. As a result, Asterisk never realizes that it has
received the ACK. Asterisk continues retransmitting a 200 OK to the phone and
the phone dutifully keeps sending an ACK in response until Asterisk has
retransmitted the maximum amount of times.
There are a couple of potential ways of solving this issue. One is to add an
Answer to your dialplan as the first priority. This way, the INVITE is
completely answered before the phone ever sends any INFO requests. Another is to
switch the phone away from using INFO to transmit DTMF. I would be willing to
bet that the other phones on your network are not using INFO for transmission of
DTMF, and so they are not experiencing the same issue.
Mark Michelson
_______________________________________________
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Posted: Fri May 22, 2009 5:32 pm Post subject: [asterisk-users] No response to our critical packet problem
for some reason (someone would have to look deeper) your SIP peer
sends ACK to 200 OK and Asterisk doesn't "get it"
so it retransmits 200 OK a couple times and then assumes there's noone there
Martin
On Fri, May 22, 2009 at 12:36 PM, James Lamanna <jlamanna@gmail.com> wrote:
Quote:
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect calls to internal office extensions (which still
go through asterisk) OR voicemail
2) The other 20+ phones in the same office on the same network have 0 problems.
Here's a SIP trace of the problem.
yyy.yyy.yyy.yyy is the outside NAT IP
xxx.xxx.xxx.xxx is the IP of my PBX
dddddddddd is the dialed phone number
sssssssssss is the source phone number
The peculiar thing is that asterisk sends an OK in response to an INVITE,
then the phone sends back an ACK, which asterisk seems to ignore
because it retransmits the OK message again
Then eventually the phone gives up and sends a BYE message.
Posted: Fri May 22, 2009 5:38 pm Post subject: [asterisk-users] No response to our critical packet problem
Quote:
I think I know what the problem is here. It's not the fault of the phone, but of
Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#',
specifically) to Asterisk. Asterisk only keeps track of the last incoming Cseq
in a dialog, so once the INFO arrives, we no longer have any memory of the Cseq
of the INVITE that the phone sent.
well then Asterisk now behaves as a poor written hand script that
handles SIP calls ...
INFO can arrive at any time when dtmfmode=info....
Martin
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Fri May 22, 2009 5:48 pm Post subject: [asterisk-users] No response to our critical packet problem
Martin wrote:
Quote:
> I think I know what the problem is here. It's not the fault of the phone, but of
> Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#',
> specifically) to Asterisk. Asterisk only keeps track of the last incoming Cseq
> in a dialog, so once the INFO arrives, we no longer have any memory of the Cseq
> of the INVITE that the phone sent.
well then Asterisk now behaves as a poor written hand script that
handles SIP calls ...
INFO can arrive at any time when dtmfmode=info....
Martin
Yes, this would be why I said that it is Asterisk's fault and provided possible
workarounds.
Thank you for your helpful and constructive criticism.
Mark Michelson
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Fri May 22, 2009 6:10 pm Post subject: [asterisk-users] No response to our critical packet problem
Quote:
Yes, this would be why I said that it is Asterisk's fault and provided possible
workarounds.
Thank you for your helpful and constructive criticism.
LOL yes you could expect now everyone to be critical about something like this.
Asterisk has been around for quite some time now (6+ years) and this
sounds like a pretty basic problem
that could cause a lot of failed calls with some SIP MTAs.
I would expect this kind of problem from an Asterisk version before 1.0.0
Martin
_______________________________________________
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Posted: Fri May 22, 2009 7:32 pm Post subject: [asterisk-users] No response to our critical packet problem
On May 22, 2009, at 3:05 PM, Martin wrote:
Quote:
> Yes, this would be why I said that it is Asterisk's fault and
> provided possible
> workarounds.
>
> Thank you for your helpful and constructive criticism.
LOL yes you could expect now everyone to be critical about something
like this.
Asterisk has been around for quite some time now (6+ years) and this
sounds like a pretty basic problem
that could cause a lot of failed calls with some SIP MTAs.
I would expect this kind of problem from an Asterisk version before
1.0.0
Martin
Things that find their way to the issue tracker tend to get repaired
if there is a reproduce-able problem. If you could include the
documentation you've made in a new issue/bug report, that would help
this get nailed down. If nobody has reported this as a bug before,
then chances are it hasn't been fixed. Additionally, as Asterisk is a
community effort it may be possible for you to create a quick patch
which if applied would solve the problem for everyone, and including
that in the issue report would help to quickly resolve the matter.
JT
--
John Todd email:jtodd@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
_______________________________________________
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Posted: Fri May 22, 2009 7:54 pm Post subject: [asterisk-users] No response to our critical packet problem
Hi Guys,
I just wanted to let you all know that you were indeed correct, it was
the SIP INFO '#'
that was causing the problem.
You'll pardon me, but I find this problem _utterly ridiculous_.
I am running asterisk v1.4.18. Are there any asterisk versions that
this is fixed on?
Thanks.
(Oh and please CC me, I'm reading in digest mode..)
-- James
On Fri, May 22, 2009 at 10:36 AM, James Lamanna <jlamanna@gmail.com> wrote:
Quote:
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect calls to internal office extensions (which still
go through asterisk) OR voicemail
2) The other 20+ phones in the same office on the same network have 0 problems.
Here's a SIP trace of the problem.
yyy.yyy.yyy.yyy is the outside NAT IP
xxx.xxx.xxx.xxx is the IP of my PBX
dddddddddd is the dialed phone number
sssssssssss is the source phone number
The peculiar thing is that asterisk sends an OK in response to an INVITE,
then the phone sends back an ACK, which asterisk seems to ignore
because it retransmits the OK message again
Then eventually the phone gives up and sends a BYE message.
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