Displaying 5 results from an estimated 5 matches for "aaws4j6i".
2015 May 11
2
"Retransmission Timeout" results in dropped calls after 32 seconds
....0/24 network (these are directly route-able so
> no
> NAT is involved). However, I have now been able to reproduce the problem
> between
> two devices directly on the 10.10.32.0/21 network as well.
>
I've gathered the log for this dialog from the SIP phone:
http://pastebin.com/aAWs4j6i
What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103,
then another INVITE is received for CSeq 103, at which point the phone
reports an error:
<0> | ERROR | receive a request with same cseq??
>From the asterisk side, it never seems to receive this OK for CSeq 10...
2015 May 12
0
"Retransmission Timeout" results in dropped calls after 32 seconds
...able so
> > no
> > NAT is involved). However, I have now been able to reproduce the problem
> > between
> > two devices directly on the 10.10.32.0/21 network as well.
> >
>
> I've gathered the log for this dialog from the SIP phone:
> http://pastebin.com/aAWs4j6i
>
> What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103,
> then another INVITE is received for CSeq 103, at which point the phone
> reports an error:
> <0> | ERROR | receive a request with same cseq??
>
> From the asterisk side, it never seems...
2015 May 13
0
"Retransmission Timeout" results in dropped calls after 32 seconds
...y #1)
INVITE (103) - re-INVITE to go to bypass mode (retry #2)
INVITE (103) - re-INVITE to go to bypass mode (retry #3)
INVITE (103) - re-INVITE to go to bypass mode (retry #4)
INVITE (103) - re-INVITE to go to bypass mode (retry #5)
Looking at the logs from the yealink phone (http://pastebin.com/aAWs4j6i),
I see a few differences:
INVITE (102) - initial call connecting
TRYING (102) - initial call connecting
RINGING (102) - initial call connecting
INVITE (102) - initial call connecting (seems like a duplicate INVITE)
RINGING (102) - initial call connecting
OK (102) - initial call connecting
ACK (102...
2015 May 11
2
"Retransmission Timeout" results in dropped calls after 32 seconds
Andrew Martin wrote:
> ----- Original Message -----
>> From: "Joshua Colp"<jcolp at digium.com>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com>
>> Sent: Monday, May 11, 2015 12:32:06 PM
>> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32
2015 May 12
2
"Retransmission Timeout" results in dropped calls after 32 seconds
Andrew Martin wrote:
<snip>
>>
> Joshua,
>
> As a mitigation for this problem, could I increase the "timerb" option in sip.conf
> to a large value, say 1 hour (instead of the default 32 seconds)? What other
> consequences would there be from this change?
I don't know if chan_sip will allow this, but if it does... it'll keep
transmitting over and