Re: [asterisk-users] Device state of SIP doesn't change
cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark, thanks for pointing this out. However, i was stuck without any success, until i tried adding my phone in static config - then it magically worked. So, i could use rtcachefriends=yes but that's something i would really like to avoid. Is this considered a bug? There's nothing in docs saying that state information is incompatible with Realtime. Regards, Atis After further discussion regarding this in #asterisk this morning, it would appear that communicating proper device state with realtime peers/ users does not work properly. I would tentatively consider this a bug since I would expect that anything that works statically should also work in realtime as well. However, since I have not done a ton of work with chan_sip myself, there could be some subtle (or not so subtle) reason why this was purposely not implemented. Sorry I can't be more authoritative on this matter. No, it is *not* a bug, it's a design. Realtime buddies are not ment to be static and get the same set of services, even if they're cached. We really need to discuss this design, since the asterisk users does not understand this and propably wants something else than what we are offering. I've sent a few mails earlier about this to asterisk-dev without getting any replies. I think the realtime dynamic caching code in chan_sip sucks, to be honest. We need static objects loaded from the realtime database. Right now it's a patchwork of patches without no good design. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Device state of SIP doesn't change
Atis Lezdins wrote: On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark, thanks for pointing this out. However, i was stuck without any success, until i tried adding my phone in static config - then it magically worked. So, i could use rtcachefriends=yes but that's something i would really like to avoid. Is this considered a bug? There's nothing in docs saying that state information is incompatible with Realtime. Regards, Atis After further discussion regarding this in #asterisk this morning, it would appear that communicating proper device state with realtime peers/users does not work properly. I would tentatively consider this a bug since I would expect that anything that works statically should also work in realtime as well. However, since I have not done a ton of work with chan_sip myself, there could be some subtle (or not so subtle) reason why this was purposely not implemented. Sorry I can't be more authoritative on this matter. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Device state of SIP doesn't change
Mark Michelson wrote: Atis Lezdins wrote: On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark, thanks for pointing this out. However, i was stuck without any success, until i tried adding my phone in static config - then it magically worked. So, i could use rtcachefriends=yes but that's something i would really like to avoid. Is this considered a bug? There's nothing in docs saying that state information is incompatible with Realtime. Regards, Atis After further discussion regarding this in #asterisk this morning, it would appear that communicating proper device state with realtime peers/users does not work properly. I would tentatively consider this a bug since I would expect that anything that works statically should also work in realtime as well. However, since I have not done a ton of work with chan_sip myself, there could be some subtle (or not so subtle) reason why this was purposely not implemented. Sorry I can't be more authoritative on this matter. Mark Michelson After some discussion on IRC, and reviewing my initial reply to you, I should clarify that proper device state reporting for realtime SIP peers does work with rtcachefriends enabled. I believe I will start up a branch soon to work out the details of getting proper device state reported for realtime SIP peers which are not cached. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Device state of SIP doesn't change
On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark, thanks for pointing this out. However, i was stuck without any success, until i tried adding my phone in static config - then it magically worked. So, i could use rtcachefriends=yes but that's something i would really like to avoid. Is this considered a bug? There's nothing in docs saying that state information is incompatible with Realtime. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Device state of SIP doesn't change
Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
