Displaying 5 results from an estimated 5 matches for "zjqzdvy3".
2015 May 11
2
"Retransmission Timeout" results in dropped calls after 32 seconds
Andrew Martin wrote:
> ----- Original Message -----
<snip>
>
> By doing a number of test calls today, I have managed to reproduce this while
> sip debugging was on, so I have that information available now as well:
> http://pastebin.com/ZJqzdvY3
>
> This was a call from 113 to 146 via a queue. Note that the asterisk server is
> at 10.10.32.251. I see the following:
> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
> SIP/2.0 180 Ringing
> SIP/2.0 180 Ringing
> SIP/2.0 200 OK
> ACK sip:146 at 10.10.32.96:5062 SIP/2.0
>...
2015 May 08
2
"Retransmission Timeout" results in dropped calls after 32 seconds
Hello,
I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All
the SIP clients are on a LAN, so no NAT is involved. I have been experiencing
an intermittent problem where a call will be successfully answered, but then
dropped by Asterisk 32 seconds after it is answered (with a "Retransmission
timeout reached on transmission" error). Here is an example of this
2015 May 11
0
"Retransmission Timeout" results in dropped calls after 32 seconds
...tin wrote:
> > ----- Original Message -----
>
> <snip>
>
> >
> > By doing a number of test calls today, I have managed to reproduce this
> > while
> > sip debugging was on, so I have that information available now as well:
> > http://pastebin.com/ZJqzdvY3
> >
> > This was a call from 113 to 146 via a queue. Note that the asterisk server
> > is
> > at 10.10.32.251. I see the following:
> > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
> > SIP/2.0 180 Ringing
> > SIP/2.0 180 Ringing
> > SIP/2.0 200 OK
>...
2015 May 11
0
"Retransmission Timeout" results in dropped calls after 32 seconds
...ow=gsm
> localnet=10.10.32.0/255.255.248.0
>
>
> [123]
> secret=111111
> host=dynamic
> type=friend
>
By doing a number of test calls today, I have managed to reproduce this while
sip debugging was on, so I have that information available now as well:
http://pastebin.com/ZJqzdvY3
This was a call from 113 to 146 via a queue. Note that the asterisk server is
at 10.10.32.251. I see the following:
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
SIP/2.0 180 Ringing
SIP/2.0 180 Ringing
SIP/2.0 200 OK
ACK sip:146 at 10.10.32.96:5062 SIP/2.0
INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
S...
2015 May 11
2
"Retransmission Timeout" results in dropped calls after 32 seconds
...ote:
>>> ----- Original Message -----
>> <snip>
>>
>>> By doing a number of test calls today, I have managed to reproduce this
>>> while
>>> sip debugging was on, so I have that information available now as well:
>>> http://pastebin.com/ZJqzdvY3
>>>
>>> This was a call from 113 to 146 via a queue. Note that the asterisk server
>>> is
>>> at 10.10.32.251. I see the following:
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>> SIP/2.0 180 Ringing
>>> SIP/2.0 180 Ringing
>>&...