VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
atis at iq-labs.net
Guest
|
 Posted: Thu Jan 17, 2008 12:36 pm Post subject: [asterisk-users] Device state of SIP doesn't change |
 |
|
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: 21168 at device
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
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835 |
|
| Back to top |
|
 |
mmichelson at digium.com
Guest
|
| Posted: Thu Jan 17, 2008 3:01 pm Post subject: [asterisk-users] Device state of SIP doesn't change |
|
|
Atis Lezdins wrote:
| Quote: | 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: 21168 at device
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 |
|
| Back to top |
|
|
atis at iq-labs.net
Guest
|
| Posted: Fri Jan 18, 2008 9:24 am Post subject: [asterisk-users] Device state of SIP doesn't change |
|
|
On 1/17/08, Mark Michelson <mmichelson at digium.com> wrote:
| Quote: | Atis Lezdins wrote:
| Quote: | 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: 21168 at device
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.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835 |
|
| Back to top |
|
|
mmichelson at digium.com
Guest
|
| Posted: Fri Jan 18, 2008 10:51 am Post subject: [asterisk-users] Device state of SIP doesn't change |
|
|
Atis Lezdins wrote:
| Quote: | On 1/17/08, Mark Michelson <mmichelson at digium.com> wrote:
| Quote: | Atis Lezdins wrote:
| Quote: | 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: 21168 at device
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 |
|
| Back to top |
|
|
mmichelson at digium.com
Guest
|
| Posted: Fri Jan 18, 2008 11:23 am Post subject: [asterisk-users] Device state of SIP doesn't change |
|
|
Mark Michelson wrote:
| Quote: | Atis Lezdins wrote:
| Quote: | On 1/17/08, Mark Michelson <mmichelson at digium.com> wrote:
| Quote: | Atis Lezdins wrote:
| Quote: | 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: 21168 at device
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 |
|
| Back to top |
|
|
oej at edvina.net
Guest
|
| Posted: Sun Jan 20, 2008 8:12 am Post subject: [asterisk-users] Device state of SIP doesn't change |
|
|
| Quote: | | Quote: | | Quote: | | Quote: |
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 |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
