[asterisk-users] Asterisk 13.13.1

Tue Jan 31 04:26:06 CST 2017

SIP packet loss is one thing, RTP packet loss is another one.
One does not necessarily imply the other though, of course, both may happen
for a common reason.

What about audio codecs ?
Is it possible to configure things so that you only have a single codec
enabled all over your system (trunks, phones, ...) ?
Do you still have audio issues with a single codec ?

2017-01-30 17:55 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>:

> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from
> here: http://downloads.asterisk.org/pub/telephony/asterisk/
> asterisk-13-current.tar.gz
>
>
>
> I continue to see errors like this:
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@
> 192.168.125.173 for seqno 109 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission 6e3dd238-911e2ac3-f1260152@
> 192.168.125.152 for seqno 103 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission
> ed38f9c8-295a9db-c23f5242 at 192.168.125.144 for seqno 103 (Critical
> Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+
> Retransmissions
>
> Packet timed out after 32000ms with no response
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission ef497d11-a81b1c00-8bfbd3bf@
> 192.168.1.244 for seqno 103 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
>
>
> Before upgrading to this new server, Asterisk version 1.8 on  CentOS 5.9
> hardware on both servers were similar in CPU, Memory
>
>
>
> Any support on this matter is appreciated!
>
>
>
> Thanks,
> Motty
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] *On Behalf Of *kambiz sharifi
> *Sent:* Saturday, January 28, 2017 5:13 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk 13.13.1
>
>
>
>
>
> 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@
> 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
>
>
> --
> _____________________________________________________________________
> -- 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
>
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