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kambiz sharifi

2017-Jan-28 13:12 UTC

[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 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 --------------
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Motty Cruz

2017-Jan-30 16:55 UTC

head link

[asterisk-users] Asterisk 13.13.1

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 at 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 at 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 at 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 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

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Doug Lytle

2017-Jan-30 17:04 UTC

head link

[asterisk-users] Asterisk 13.13.1

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.cruz at gmail.com
wrote:
>>> 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 at
192.168.125.173 for seqno 109 (Critical Request) -- See >>>
>>>
Firewall?

Doug

Michael Maier

2017-Jan-30 17:14 UTC

head link

[asterisk-users] Asterisk 13.13.1

On 01/30/2017 at 05:55 PM Motty Cruz wrote:> 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 at
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 at
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 at
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!
Did you setup tcpdump (behind the machine) to see, if the packets really
leave the machine? Can you see any answer?


Regards,
Michael

Olivier

2017-Jan-31 10:26 UTC

head link

[asterisk-users] Asterisk 13.13.1

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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