Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
Hi Nothing will stop the behaviour you are seeing. A SIP reload will clear the realtime cache thus stopping the asterisk server knowing where the realtime sip endpoint is until the endpoint re-registers. The question here is, why are you doing SIP reloads? Once you are using RealTime architecture for SIP, sip reloads become unnecessary unless you are making modifications to the general section of your sip.conf and why would you need to do that regularly? Regards Ish On Wed, 2012-02-15 at 12:52 +0530, DHAVAL INDRODIYA wrote: i tried it and it wont work with rtcachefriend=yes On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson jmr.richard...@gmail.com wrote: 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 The first thing I would try is 'rtcachefriends=yes', that should do it. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
i tried it and it wont work with rtcachefriend=yes On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson jmr.richard...@gmail.comwrote: 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 The first thing I would try is 'rtcachefriends=yes', that should do it. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.
Hi To the best of my understanding this is the correct behaviour. When you add a peer to the database and a device configured for that peer registers, it enters that peer into the RealTime cache. When you do a sip reload you fully clear that RealTime cache so the asterisk process will lose knowledge of that peer until it registers again and gets re entered into the RealTime cache, which most SIP phones are set to do after a number of minutes. The real question here is, if you are using RealTime architecture for your peers, why are you doing a sip reload? On Fri, 2012-02-10 at 10:55 +0530, DHAVAL INDRODIYA wrote: nobody facing any issue with this or nobody using real time architecture On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA dhaval.it01...@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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing
Re: [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.it01...@gmail.com Sent: Friday, February 10, 2012 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@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.it01...@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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
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 The first thing I would try is 'rtcachefriends=yes', that should do it. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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.it01...@gmail.comwrote: 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
