Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!
I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf
[1091]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1091>
disallow=all
allow=ulaw
allow=alaw
username=1091
secret=XXXXX
dtmfmode=rfc2833
host=dynamic
mailbox=10091 at default
nat=force_rport,comedia
canreinvite=no
extensions.conf
exten => 1091,hint,SIP/${EXTEN}
exten => 1091,1,Dial(SIP/${EXTEN},15,t)
exten => 1091,2,Voicemail(${EXTEN}@default,u)
exten => 1091,102,Voicemail(${EXTEN}@default,b)
exten => 1091,103,Hangup
[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:
Retransmission timeout reached on transmission
7c803889-63e1b3fe-c2b5ef77 at 192.168.0.191 for seqno 156 (Critical Request) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
any ideas?
Thanks!
Motty
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What did you exactly upgade ? Asterisk only ? Asterisk and OS ? How did you installed Asterisk 1.8 and 13 ? From source or from package ? I would be curious to see what would happen after downgrading back to 1.8. 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>:> Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are > starting to complaint about packets loss, conversations are choppy! > > > > > I don?t even know where to start looking! Choppy conversations happened > within users. I am using sip.conf > > > > [1091] > > type=friend > > context=sip-phone > > call-limit=2 > > trustrpid=no > > callerid="dev1" <1091> > > disallow=all > > allow=ulaw > > allow=alaw > > username=1091 > > secret=XXXXX > > dtmfmode=rfc2833 > > host=dynamic > > mailbox=10091 at default > > nat=force_rport,comedia > > canreinvite=no > > > > extensions.conf > > exten => 1091,hint,SIP/${EXTEN} > > exten => 1091,1,Dial(SIP/${EXTEN},15,t) > > exten => 1091,2,Voicemail(${EXTEN}@default,u) > > exten => 1091,102,Voicemail(${EXTEN}@default,b) > > exten => 1091,103,Hangup > > > > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: > > Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@ > 192.168.0.191 for seqno 156 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > > > any ideas? > > > > Thanks! > > Motty > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170125/60b7d61d/attachment.html>
On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote:> What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>: > > Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are > starting to complaint about packets loss, conversations are choppy! > > > > > PkEI don?t even know where to start looking! Choppy conversations happened > within users. I am using sip.conf > > > > [1091] > > type=friend > > context=sip-phone > > call-limit=2 > > trustrpid=no > > callerid="dev1" <1091> > > disallow=all > > allow=ulaw > > allow=alaw > > username=1091 > > secret=XXXXX > > dtmfmode=rfc2833 > > host=dynamic > > mailbox=10091 at default > > nat=force_rport,comedia > > canreinvite=no > > > > extensions.conf > > exten => 1091,hint,SIP/${EXTEN} > > exten => 1091,1,Dial(SIP/${EXTEN},15,t) > > exten => 1091,2,Voicemail(${EXTEN}@default,u) > > exten => 1091,102,Voicemail(${EXTEN}@default,b) > > exten => 1091,103,Hangup > > > > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: > > Retransmission timeout reached on transmission > 7c803889-63e1b3fe-c2b5ef77 at 192.168.0.191 for seqno 156 (Critical Request) > -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > > > any ideas? > > > > Thanks! > > Motty > > -- > > > _____________________________________________________________________ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > > > > > New to Asterisk? Start here: > > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170128/7b9dc1f8/attachment.html>