[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.
BryantZ at zktech.com
Fri Feb 10 06:21:01 CST 2012
I see this on some peers every time I do a sip reload and I am not using
real-time. I use qualify and every time a sip reload occurs the device goes
unreachable. I have shortend the register time to 5 min so the device
comes back with-in about two min but it is very annonying to me and my
user. I have tracked my issue back to cusomters using netgear routers. If
they replace the device the issue goes away. On netgear routers we have
found we have to shut of SIP AGL to get them to register right but this
quark won't go away. Maybe your issue is endpoint releated as well?
From: "DHAVAL INDRODIYA" <dhaval.it01034 at gmail.com>
Sent: Friday, February 10, 2012 12:22 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Asterisk SIP Realtime Architecture
nobody facing any issue with this or nobody using real time architecture
On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA
<dhaval.it01034 at gmail.com> wrote:
I am facing an issue with Peer registration in my asterisk server .
I am using asterisk version 22.214.171.124 and using SIP real-time
architecture.when i am doing registration it registered fine on asterisk
as peer is available in Database.
But now i am doing 'sip reload' or 'reload' due to some reason my peer
registration is going out and i cannot able to call that peer even though
in SIP client it shows me 'registered'.
Can any body elaborate on this issue which settings i need to put in
I also tried to follow this patch
https://issues.asterisk.org/view.php?id=14196 But it allready applied in
code base so why it wont work?
Here is my sip.conf settings.
context=from-internal ; Default context for incoming cal
callcounter = yes
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
pedantic=yes ; Enable slow, pedantic checking for Pingtel
tos=184 ; Set IP QoS to either a keyword or numeric val
tos_sip=cs3 ; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
maxexpiry=3600 ; Max length of incoming registration we allow
defaultexpiry=120 ; Default length of incoming/outoing registration
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
language=en ; Default language setting for all
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP
useragent=dhaval ; Allows you to change the user agent string
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default:
And here is DB fields snapshots.
useragent: CSipSimple r1133 / b
Kindly help me to resolve this.
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the asterisk-users