Re: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

2017-04-04 Thread Motty Cruz 
Hello, I am trying to match SRC in CSV file to clid - Caller ID to user
extension number for stats purposes, however, in CSV file the SRC is the
company number as set in Extensions.conf 

 

exten => _7XXX,1,Set(CALLERID(number)="Company Inc" <3788001800>)

exten => _7XXX,2,Dial(SIP/voip1/13781${EXTEN:1},80)

exten => _7XXX,n,Congestion()

exten => _7XXX,n,Hangup()

 

how would I change it? I have look in cdr.conf and logger.conf

 

Thanks, 

 

From: Motty Cruz [mailto:motty.c...@gmail.com] 
Sent: Monday, April 03, 2017 3:52 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: motty.c...@gmail.com
Subject: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID
as CallerID

 

Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to  log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number is 2318001800 which is the main company's number set in
Extensions.conf. 

 



2318001800

78079745

phones

"ITadmin" <2318001800>

SIP/4007-00015c0a

SIP/voip1-00015c0b

Dial

SIP/voip1/78079745,80

4/3/2017 15:30

4/3/2017 15:31

2

0

NO ANSWER

DOCUMENTATION

1.49E+09




 

 

Logger.conf

[general]

dateformat=%F %T

;

; Customize the display of debug message time stamps

; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)

;

; see strftime(3) Linux manual for format specifiers.  Note that there is
also

; a fractional second parameter which may be used in this field.  Use %1q

; for tenths, %2q for hundredths, etc.

;

;dateformat=%F %T   ; ISO 8601 date format

;dateformat=%F %T.%3q   ; with milliseconds

dateformat = %F %T.%3q   ; ISO 8601 date format with milliseconds

;

;

; This makes Asterisk write callids to log messages

; (defaults to yes)

use_callids = yes

 

 

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[asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

2017-04-03 Thread Motty Cruz 
Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to  log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number is 2318001800 which is the main company's number set in
Extensions.conf. 

 



2318001800

78079745

phones

"ITadmin" <2318001800>

SIP/4007-00015c0a

SIP/voip1-00015c0b

Dial

SIP/voip1/78079745,80

4/3/2017 15:30

4/3/2017 15:31

2

0

NO ANSWER

DOCUMENTATION

1.49E+09




 

 

Logger.conf

[general]

dateformat=%F %T

;

; Customize the display of debug message time stamps

; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)

;

; see strftime(3) Linux manual for format specifiers.  Note that there is
also

; a fractional second parameter which may be used in this field.  Use %1q

; for tenths, %2q for hundredths, etc.

;

;dateformat=%F %T   ; ISO 8601 date format

;dateformat=%F %T.%3q   ; with milliseconds

dateformat = %F %T.%3q   ; ISO 8601 date format with milliseconds

;

;

; This makes Asterisk write callids to log messages

; (defaults to yes)

use_callids = yes

 

 

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[asterisk-users] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)

2017-02-02 Thread Motty Cruz 
Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error: 

 

-- SIP/voipeer-084b redirecting info has changed, passing it to
SIP/1007-084a

-- SIP/voipeer-084b is busy

  == Everyone is busy/congested at this time (1:1/0/0)

-- Timeout on SIP/1007-084a

-- Executing [t@phones:1] Playback("SIP/1007-084a", "goodbye") in
new stack

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

--  Playing 'goodbye.slin' (language 'en')

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

-- Executing [t@phones:2] Hangup("SIP/1007-084a", "") in new stack

 

Sip.conf 

[1007]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1007>
disallow=all
allow=ulaw
allow=alaw
username=1007
secret=X
dtmfmode=rfc2833
host=dynamic
mailbox=1007@default
nat=force_rport,comedia

 

Is it a codec issue? Or missed configuration? Asterisk does not know how to
translate busy signal. 

Your help is appreciated!

Thanks,

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Re: [asterisk-users] Asterisk 13.13.1

2017-01-31 Thread Olivier 
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.c...@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@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-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@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.4...@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.c...@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=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@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 UNSUBSCR

Re: [asterisk-users] Asterisk 13.13.1

2017-01-30 Thread Ron Wheeler 

CentOS 7 uses firewalld to control TCP amd UDP access.

The iptables configuration will be overwritten and dynamically changed 
by Firewalld so don't count on the old practice of manipulating iptables 
directly.


I recently moved our Asterisk from an old CentOS to CentOS 7 running 
FreePBX 14.0.1.beta2.


You can add a firewalld service yp /etc/firewalld/services like mine.
[root@firewall0 services]# cat Asterisk.xml


  asterisk
  Asterisk PBX
  
  
  
  
  


You then permit this service in your interface (zones) as a service
 

I also added a rule to get some logging on the Asterisk ports while 
getting things up and running.

  



  
  


I did this all on my exterior firewall which is also a CentOS 7 system.
On the Asterisk server, I do not block anything which is not a best 
practice but the entire internal network is very small and I consider it 
to be secure.


You (and I) should control the interface using Firewalld with the same 
service and zone specifications.






On 30/01/2017 12:13 PM, Motty Cruz wrote:

I thought it was a firewall issues. I disabled IP Tables & Selinux, but the
problem persist! I have not made changes on our firewall since the upgrade!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1


On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@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@192.168.125.173 for seqno 109 (Critical Request)
-- See >>> >>>

Firewall?

Doug




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Asterisk 13.13.1

2017-01-30 Thread Michael Maier 
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@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@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!

Did you setup tcpdump (behind the machine) to see, if the packets really
leave the machine? Can you see any answer?


Regards,
Michael

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Re: [asterisk-users] Asterisk 13.13.1

2017-01-30 Thread Motty Cruz 
I thought it was a firewall issues. I disabled IP Tables & Selinux, but the
problem persist! I have not made changes on our firewall since the upgrade!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@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@192.168.125.173 for seqno 109 (Critical Request)
-- See >>> >>> 

Firewall?

Doug

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Re: [asterisk-users] Asterisk 13.13.1

2017-01-30 Thread Doug Lytle 
>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@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@192.168.125.173 for seqno 109 (Critical Request) 
>>> -- See >>> >>> 

Firewall?

Doug

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Re: [asterisk-users] Asterisk 13.13.1

2017-01-30 Thread Motty Cruz 
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@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-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@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.4...@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.c...@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=X

dtmfmode=rfc2833

host=dynamic

mailbox=10091@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


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Re: [asterisk-users] Asterisk 13.13.1

2017-01-28 Thread kambiz sharifi 
On Wed, Jan 25, 2017 at 16:00 Olivier  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 :
>
> 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=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@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
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Re: [asterisk-users] Asterisk 13.13.1

2017-01-25 Thread Olivier 
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 :

> 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=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@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
>
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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 13.13.1

2017-01-24 Thread Motty Cruz 
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=X

dtmfmode=rfc2833

host=dynamic

mailbox=10091@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

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