DHAVAL INDRODIYA
2012-Feb-09 05:24 UTC
[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.
Hi Group. I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 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 sip.conf. 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. [general] context=from-internal ; Default context for incoming cal rtcachefriends=no rtupdate=yes rtautoclear=yes rtsavesysname=yes callcounter = yes callevents=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 preferred_codec_only=yes disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw insecure=invite language=en ; Default language setting for all users/peers rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=dhaval ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 qualify=yes nat=yes ;canreinvite=yes directmedia=yes directrtpsetup=yes And here is DB fields snapshots. id: 1 name: 201 ipaddr: 172.18.100.243 port: 53624 regseconds: 1328716180 defaultuser: 201 fullcontact: NULL regserver: dhaval useragent: CSipSimple r1133 / b lastms: 554 host: dynamic type: friend context: from-internal permit: NULL deny: NULL secret: 201 md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: yes nat: NULL allow: ulaw disallow: g729 insecure: invite callerid: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL Kindly help me to resolve this. Thanks Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120209/e489046d/attachment.htm>
DHAVAL INDRODIYA
2012-Feb-10 05:25 UTC
[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.
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:> Hi Group. > > I am facing an issue with Peer registration in my asterisk server . > > I am using asterisk version 1.8.5.0 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 > sip.conf. > > 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. > > > [general] > context=from-internal ; Default context for incoming cal > rtcachefriends=no > rtupdate=yes > rtautoclear=yes > rtsavesysname=yes > callcounter = yes > callevents=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 > preferred_codec_only=yes > disallow=all ; First disallow all codecs > allow=ulaw ; Allow codecs in order of preference > allow=alaw > insecure=invite > language=en ; Default language setting for all > users/peers > rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP > activity > useragent=dhaval ; Allows you to change the user agent string > dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. > Default: rfc2833 > qualify=yes > nat=yes > ;canreinvite=yes > directmedia=yes > directrtpsetup=yes > > And here is DB fields snapshots. > > id: 1 > name: 201 > ipaddr: 172.18.100.243 > port: 53624 > regseconds: 1328716180 > defaultuser: 201 > fullcontact: NULL > regserver: dhaval > useragent: CSipSimple r1133 / b > lastms: 554 > host: dynamic > type: friend > context: from-internal > permit: NULL > deny: NULL > secret: 201 > md5secret: NULL > remotesecret: NULL > transport: NULL > dtmfmode: NULL > directmedia: yes > nat: NULL > allow: ulaw > disallow: g729 > insecure: invite > callerid: NULL > rfc2833compensate: NULL > mailbox: NULL > session-timers: NULL > session-expires: NULL > session-minse: NULL > session-refresher: NULL > > > Kindly help me to resolve this. > > Thanks > Dhaval > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120210/55303297/attachment.htm>
Bryant Zimmerman
2012-Feb-10 12:21 UTC
[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.
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? Bryant ---------------------------------------- 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 Issue/Bug. 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: Hi Group. I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 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 sip.conf. 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. [general] context=from-internal ; Default context for incoming cal rtcachefriends=no rtupdate=yes rtautoclear=yes rtsavesysname=yes callcounter = yes callevents=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 preferred_codec_only=yes disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw insecure=invite language=en ; Default language setting for all users/peers rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=dhaval ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 qualify=yes nat=yes ;canreinvite=yes directmedia=yes directrtpsetup=yes And here is DB fields snapshots. id: 1 name: 201 ipaddr: 172.18.100.243 port: 53624 regseconds: 1328716180 defaultuser: 201 fullcontact: NULL regserver: dhaval useragent: CSipSimple r1133 / b lastms: 554 host: dynamic type: friend context: from-internal permit: NULL deny: NULL secret: 201 md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: yes nat: NULL allow: ulaw disallow: g729 insecure: invite callerid: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL Kindly help me to resolve this. Thanks Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120210/f4adac49/attachment.htm>