<div dir="ltr">hi,<div><br></div><div>Do you edit your </div><div><br></div><div>voicemail.conf?</div><div>[default]</div><div><font color="#333333" face="consolas, liberation mono, menlo, courier, monospace"><span style="font-size:12px;white-space:pre;background-color:rgb(166,243,166)">1091=(number to access your voicemail in your phone ex: 1234)</span></font><br></div><div><font color="#333333" face="consolas, liberation mono, menlo, courier, monospace"><span style="font-size:12px;white-space:pre;background-color:rgb(166,243,166)"><br></span></font></div><div><font color="#333333" face="consolas, liberation mono, menlo, courier, monospace"><span style="font-size:12px;white-space:pre;background-color:rgb(166,243,166)"><br></span></font></div><div><font color="#333333" face="consolas, liberation mono, menlo, courier, monospace"><span style="font-size:12px;white-space:pre;background-color:rgb(166,243,166)"><br></span></font></div><div><font color="#333333" face="consolas, liberation mono, menlo, courier, monospace"><span style="font-size:12px;white-space:pre;background-color:rgb(166,243,166)"><br></span></font></div></div><div class="gmail_extra"><br><div class="gmail_quote">2017-01-25 16:00 GMT-02:00  <span dir="ltr"><<a href="mailto:asterisk-users-request@lists.digium.com" target="_blank">asterisk-users-request@lists.digium.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send asterisk-users mailing list submissions to<br>
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Today's Topics:<br>
<br>
   1.  Asterisk 13.13.1 (Motty Cruz)<br>
   2. Re: Asterisk 13.13.1 (Olivier)<br>
<br>
<br>
------------------------------<wbr>------------------------------<wbr>----------<br>
<br>
Message: 1<br>
Date: Tue, 24 Jan 2017 12:03:05 -0800<br>
From: "Motty Cruz" <<a href="mailto:motty.cruz@gmail.com">motty.cruz@gmail.com</a>><br>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"<br>
        <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.<wbr>com</a>><br>
Subject: [asterisk-users]  Asterisk 13.13.1<br>
Message-ID: <<a href="mailto:5887b2fb.86c5620a.8e94a.d33d@mx.google.com">5887b2fb.86c5620a.8e94a.d33d@<wbr>mx.google.com</a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are<br>
starting to complaint about packets loss, conversations are choppy!<br>
<br>
<br>
<br>
<br>
I don't even know where to start looking! Choppy conversations happened<br>
within users. I am using sip.conf<br>
<br>
<br>
<br>
[1091]<br>
<br>
type=friend<br>
<br>
context=sip-phone<br>
<br>
call-limit=2<br>
<br>
trustrpid=no<br>
<br>
callerid="dev1" <1091><br>
<br>
disallow=all<br>
<br>
allow=ulaw<br>
<br>
allow=alaw<br>
<br>
username=1091<br>
<br>
secret=XXXXX<br>
<br>
dtmfmode=rfc2833<br>
<br>
host=dynamic<br>
<br>
mailbox=10091@default<br>
<br>
nat=force_rport,comedia<br>
<br>
canreinvite=no<br>
<br>
<br>
<br>
extensions.conf<br>
<br>
exten => 1091,hint,SIP/${EXTEN}<br>
<br>
exten => 1091,1,Dial(SIP/${EXTEN},15,t)<br>
<br>
exten => 1091,2,Voicemail(${EXTEN}@<wbr>default,u)<br>
<br>
exten => 1091,102,Voicemail(${EXTEN}@<wbr>default,b)<br>
<br>
exten => 1091,103,Hangup<br>
<br>
<br>
<br>
[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:<br>
<br>
Retransmission timeout reached on transmission<br>
<a href="mailto:7c803889-63e1b3fe-c2b5ef77@192.168.0.191">7c803889-63e1b3fe-c2b5ef77@<wbr>192.168.0.191</a> for seqno 156 (Critical Request) --<br>
See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" rel="noreferrer" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/SIP+<wbr>Retransmissions</a><br>
<br>
Packet timed out after 32000ms with no response<br>
<br>
<br>
<br>
any ideas?<br>
<br>
<br>
<br>
Thanks!<br>
<br>
Motty<br>
<br>
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<br>
Message: 2<br>
Date: Wed, 25 Jan 2017 13:30:00 +0100<br>
From: Olivier <<a href="mailto:oza.4h07@gmail.com">oza.4h07@gmail.com</a>><br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
        <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.<wbr>com</a>><br>
Subject: Re: [asterisk-users] Asterisk 13.13.1<br>
Message-ID:<br>
        <<a href="mailto:CAPeT9jjqkD3nWJG5j9vq8sDgsKmiQpGi0ENDKXYhwfGaVzTv2w@mail.gmail.com">CAPeT9jjqkD3nWJG5j9vq8sDgsKmi<wbr>QpGi0ENDKXYhwfGaVzTv2w@mail.<wbr>gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
What did you exactly upgade ? Asterisk only ? Asterisk and OS ?<br>
How did you installed Asterisk 1.8 and 13 ? From source or from package ?<br>
<br>
I would be curious to see what would happen after downgrading back to 1.8.<br>
<br>
2017-01-24 21:03 GMT+01:00 Motty Cruz <<a href="mailto:motty.cruz@gmail.com">motty.cruz@gmail.com</a>>:<br>
<br>
> Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are<br>
> starting to complaint about packets loss, conversations are choppy!<br>
><br>
><br>
><br>
><br>
> I don?t even know where to start looking! Choppy conversations happened<br>
> within users. I am using sip.conf<br>
><br>
><br>
><br>
> [1091]<br>
><br>
> type=friend<br>
><br>
> context=sip-phone<br>
><br>
> call-limit=2<br>
><br>
> trustrpid=no<br>
><br>
> callerid="dev1" <1091><br>
><br>
> disallow=all<br>
><br>
> allow=ulaw<br>
><br>
> allow=alaw<br>
><br>
> username=1091<br>
><br>
> secret=XXXXX<br>
><br>
> dtmfmode=rfc2833<br>
><br>
> host=dynamic<br>
><br>
> mailbox=10091@default<br>
><br>
> nat=force_rport,comedia<br>
><br>
> canreinvite=no<br>
><br>
><br>
><br>
> extensions.conf<br>
><br>
> exten => 1091,hint,SIP/${EXTEN}<br>
><br>
> exten => 1091,1,Dial(SIP/${EXTEN},15,t)<br>
><br>
> exten => 1091,2,Voicemail(${EXTEN}@<wbr>default,u)<br>
><br>
> exten => 1091,102,Voicemail(${EXTEN}@<wbr>default,b)<br>
><br>
> exten => 1091,103,Hangup<br>
><br>
><br>
><br>
> [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:<br>
><br>
> Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@<br>
> <a href="tel:192.168.0.191" value="+551921680191">192.168.0.191</a> for seqno 156 (Critical Request) -- See<br>
> <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" rel="noreferrer" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/SIP+<wbr>Retransmissions</a><br>
><br>
> Packet timed out after 32000ms with no response<br>
><br>
><br>
><br>
> any ideas?<br>
><br>
><br>
><br>
> Thanks!<br>
><br>
> Motty<br>
><br>
> --<br>
> ______________________________<wbr>______________________________<wbr>_________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br>
><br>
> Check out the new Asterisk community forum at: <a href="https://community.asterisk" rel="noreferrer" target="_blank">https://community.asterisk</a>.<br>
> org/<br>
><br>
> New to Asterisk? Start here:<br>
>       <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/Getting+<wbr>Started</a><br>
><br>
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New to Asterisk? Start here:<br>
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End of asterisk-users Digest, Vol 150, Issue 17<br>
******************************<wbr>*****************<br>
</blockquote></div><br></div>