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Frederic Jean

2006-Mar-21 14:51 UTC

[Asterisk-Users] Fw: anybody has SIP realtime working ?

Hello,

I am just asking this because I am note sure if the problem
is on my side or not, I saw some comments on SIP realtime
today so I was wondering, has anybody has SIP realtime working
with a softfone ?

If yes, please confirm, that would give me a light.
My previous message to the list is below.

Thanks.

Frederic


----- Original Message ----- 
From: Frederic Jean 
To: asterisk-users@lists.digium.com 
Sent: Tuesday, March 21, 2006 14:21
Subject: SIP Realtime 1.2.5 and Username/auth name mismatch ?



Hello, 

I installed 1.2.5 and realtime SIP. The connection to the DB is OK
because I can get the values from the CLI.

Here are my 3 different cases:

1- If I put an unexisting user, I get 404 and I am not able to dial.
2- If I check "Disable registration" within Firefly it does not
register but I am able to dial a destination (...)
3- If I leave registration ON, I get the 404 message but I am not able to dial a
destination....

This is weird, anyone had this before ??  :- )

Thanks in advance !
Frederic


Mar 21 14:11:36 NOTICE[894]: chan_sip.c:10854 handle_request_register:
Registration from
'"1111"<sip:1111@192.168.1.2:5060;transport=udp>' failed
for '192.168.1.5' - Username/auth name mismatch

SNET-PBX*CLI> realtime mysql status
Connected to CCARDS@127.0.0.1, port 3306 with username asterisk for 1 minutes,
35 seconds.

SNET-PBX*CLI> realtime load sipusers username 1111
                   Column Name  Column Value
          --------------------  --------------------
                            id  1
                          name  1111
                   accountcode  1111
                      callerid  1111
                   canreinvite  no
                       context  internal
                     defaultip  0.0.0.0
                          host  dynamic
                      insecure  very
                      language  br
                           nat  yes
                          port  0
                       qualify  no
                        secret  2222
                          type  friend
                      username  1111
                      disallow  all
                         allow  g729
                         allow  ilbc
                         allow  gsm
                         allow  ulaw
                         allow  alaw
                    regseconds  0
                        ipaddr  0.0.0.0
                cancallforward  yes


+----------------+--------------+------+-----+-------------------------+----------------+
| Field          | Type         | Null | Key | Default                 | Extra  
|
+----------------+--------------+------+-----+-------------------------+----------------+
| id             | int(11)      |      | PRI | NULL                    |
auto_increment |
| name           | varchar(80)  |      | UNI |                         |        
|
| accountcode    | varchar(20)  | YES  |     | NULL                    |        
|
| amaflags       | varchar(13)  | YES  |     | NULL                    |        
|
| callgroup      | varchar(10)  | YES  |     | NULL                    |        
|
| callerid       | varchar(80)  | YES  |     | NULL                    |        
|
| canreinvite    | char(3)      | YES  |     | yes                     |        
|
| context        | varchar(80)  | YES  |     | NULL                    |        
|
| defaultip      | varchar(15)  | YES  |     | NULL                    |        
|
| dtmfmode       | varchar(7)   | YES  |     | NULL                    |        
|
| fromuser       | varchar(80)  | YES  |     | NULL                    |        
|
| fromdomain     | varchar(80)  | YES  |     | NULL                    |        
|
| fullcontact    | varchar(80)  | YES  |     | NULL                    |        
|
| host           | varchar(31)  |      |     |                         |        
|
| insecure       | varchar(4)   | YES  |     | NULL                    |        
|
| language       | char(2)      | YES  |     | NULL                    |        
|
| mailbox        | varchar(50)  | YES  |     | NULL                    |        
|
| md5secret      | varchar(80)  | YES  |     | NULL                    |        
|
| nat            | varchar(5)   |      |     | no                      |        
|
| deny           | varchar(95)  | YES  |     | NULL                    |        
|
| permit         | varchar(95)  | YES  |     | NULL                    |        
|
| mask           | varchar(95)  | YES  |     | NULL                    |        
|
| pickupgroup    | varchar(10)  | YES  |     | NULL                    |        
|
| port           | varchar(5)   |      |     |                         |        
|
| qualify        | char(3)      | YES  |     | NULL                    |        
|
| restrictcid    | char(1)      | YES  |     | NULL                    |        
|
| rtptimeout     | char(3)      | YES  |     | NULL                    |        
|
| rtpholdtimeout | char(3)      | YES  |     | NULL                    |        
|
| secret         | varchar(80)  | YES  |     | NULL                    |        
|
| type           | varchar(6)   |      |     | friend                  |        
|
| username       | varchar(80)  |      |     |                         |        
|
| disallow       | varchar(100) | YES  |     | all                     |        
|
| allow          | varchar(100) | YES  |     | g729;ilbc;gsm;ulaw;alaw |        
|
| musiconhold    | varchar(100) | YES  |     | NULL                    |        
|
| regseconds     | int(11)      |      |     | 0                       |        
|
| ipaddr         | varchar(15)  |      |     |                         |        
|
| regexten       | varchar(80)  |      |     |                         |        
|
| cancallforward | char(3)      | YES  |     | yes                     |        
|
| setvar         | varchar(100) |      |     |                         |        
|
+----------------+--------------+------+-----+-------------------------+----------------+
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Aaron Daniel

2006-Mar-21 15:34 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

We have a few soft-phones working with the realtime setup.  What does 
the configuration of your phone look like?

Aaron

On Tue, 21 Mar 2006, Frederic 
Jean wrote:
> Hello,
>
> I am just asking this because I am note sure if the problem
> is on my side or not, I saw some comments on SIP realtime
> today so I was wondering, has anybody has SIP realtime working
> with a softfone ?
>
> If yes, please confirm, that would give me a light.
> My previous message to the list is below.
>
> Thanks.
>
> Frederic
>
>
> ----- Original Message -----
> From: Frederic Jean
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, March 21, 2006 14:21
> Subject: SIP Realtime 1.2.5 and Username/auth name mismatch ?
>
>
>
> Hello,
>
> I installed 1.2.5 and realtime SIP. The connection to the DB is OK
> because I can get the values from the CLI.
>
> Here are my 3 different cases:
>
> 1- If I put an unexisting user, I get 404 and I am not able to dial.
> 2- If I check "Disable registration" within Firefly it does not
register but I am able to dial a destination (...)
> 3- If I leave registration ON, I get the 404 message but I am not able to
dial a destination....
>
> This is weird, anyone had this before ??  :- )
>
> Thanks in advance !
> Frederic
>
>
> Mar 21 14:11:36 NOTICE[894]: chan_sip.c:10854 handle_request_register:
Registration from
'"1111"<sip:1111@192.168.1.2:5060;transport=udp>' failed
for '192.168.1.5' - Username/auth name mismatch
>
> SNET-PBX*CLI> realtime mysql status
> Connected to CCARDS@127.0.0.1, port 3306 with username asterisk for 1
minutes, 35 seconds.
>
> SNET-PBX*CLI> realtime load sipusers username 1111
>                   Column Name  Column Value
>          --------------------  --------------------
>                            id  1
>                          name  1111
>                   accountcode  1111
>                      callerid  1111
>                   canreinvite  no
>                       context  internal
>                     defaultip  0.0.0.0
>                          host  dynamic
>                      insecure  very
>                      language  br
>                           nat  yes
>                          port  0
>                       qualify  no
>                        secret  2222
>                          type  friend
>                      username  1111
>                      disallow  all
>                         allow  g729
>                         allow  ilbc
>                         allow  gsm
>                         allow  ulaw
>                         allow  alaw
>                    regseconds  0
>                        ipaddr  0.0.0.0
>                cancallforward  yes
>
>
>
+----------------+--------------+------+-----+-------------------------+----------------+
> | Field          | Type         | Null | Key | Default                 |
Extra          |
>
+----------------+--------------+------+-----+-------------------------+----------------+
> | id             | int(11)      |      | PRI | NULL                    |
auto_increment |
> | name           | varchar(80)  |      | UNI |                         |   
|
> | accountcode    | varchar(20)  | YES  |     | NULL                    |   
|
> | amaflags       | varchar(13)  | YES  |     | NULL                    |   
|
> | callgroup      | varchar(10)  | YES  |     | NULL                    |   
|
> | callerid       | varchar(80)  | YES  |     | NULL                    |   
|
> | canreinvite    | char(3)      | YES  |     | yes                     |   
|
> | context        | varchar(80)  | YES  |     | NULL                    |   
|
> | defaultip      | varchar(15)  | YES  |     | NULL                    |   
|
> | dtmfmode       | varchar(7)   | YES  |     | NULL                    |   
|
> | fromuser       | varchar(80)  | YES  |     | NULL                    |   
|
> | fromdomain     | varchar(80)  | YES  |     | NULL                    |   
|
> | fullcontact    | varchar(80)  | YES  |     | NULL                    |   
|
> | host           | varchar(31)  |      |     |                         |   
|
> | insecure       | varchar(4)   | YES  |     | NULL                    |   
|
> | language       | char(2)      | YES  |     | NULL                    |   
|
> | mailbox        | varchar(50)  | YES  |     | NULL                    |   
|
> | md5secret      | varchar(80)  | YES  |     | NULL                    |   
|
> | nat            | varchar(5)   |      |     | no                      |   
|
> | deny           | varchar(95)  | YES  |     | NULL                    |   
|
> | permit         | varchar(95)  | YES  |     | NULL                    |   
|
> | mask           | varchar(95)  | YES  |     | NULL                    |   
|
> | pickupgroup    | varchar(10)  | YES  |     | NULL                    |   
|
> | port           | varchar(5)   |      |     |                         |   
|
> | qualify        | char(3)      | YES  |     | NULL                    |   
|
> | restrictcid    | char(1)      | YES  |     | NULL                    |   
|
> | rtptimeout     | char(3)      | YES  |     | NULL                    |   
|
> | rtpholdtimeout | char(3)      | YES  |     | NULL                    |   
|
> | secret         | varchar(80)  | YES  |     | NULL                    |   
|
> | type           | varchar(6)   |      |     | friend                  |   
|
> | username       | varchar(80)  |      |     |                         |   
|
> | disallow       | varchar(100) | YES  |     | all                     |   
|
> | allow          | varchar(100) | YES  |     | g729;ilbc;gsm;ulaw;alaw |   
|
> | musiconhold    | varchar(100) | YES  |     | NULL                    |   
|
> | regseconds     | int(11)      |      |     | 0                       |   
|
> | ipaddr         | varchar(15)  |      |     |                         |   
|
> | regexten       | varchar(80)  |      |     |                         |   
|
> | cancallforward | char(3)      | YES  |     | yes                     |   
|
> | setvar         | varchar(100) |      |     |                         |   
|
>
+----------------+--------------+------+-----+-------------------------+----------------+
>
-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
amdtech@shsu.edu
(936) 294-4198

JR Richardson

2006-Mar-21 16:05 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

> Message: 16
> Date: Tue, 21 Mar 2006 18:51:29 -0300
> From: "Frederic Jean" <fjean@sunnetgroup.net>
> Subject: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> To: <asterisk-users@lists.digium.com>
> Message-ID: <0f7601c64d31$a05c5620$0501a8c0@snet01>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Hello,
> 
> I am just asking this because I am note sure if the problem
> is on my side or not, I saw some comments on SIP realtime
> today so I was wondering, has anybody has SIP realtime working
> with a softfone ?
> 
> If yes, please confirm, that would give me a light.
> My previous message to the list is below.
> 
> Thanks.
> 
> Frederic

Yes,

I have realtime working with SIP Cisco and Polycom hard phones, and DIAX
Softphones for IAX (pretty much same config as SIP.  Sip and Iax realtime works
fine for me, backing into a MySQL database.  If the softphone is giving you a
problem, try another sofphone, there are a lot if free ones to try.  There is no
reason why a soft phone would not work and a hard phone would work, except
configuration or SIP stack implementation on the soft phone.

JR Richardson
Engineering for the Masses

Douglas Garstang

2006-Mar-21 16:12 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

I had to drop realtime with sip users. If you do a reload or a restart, you lose
all the sip peer information (even with rtcachefriends=yes). That just
wasn't acceptable for us.
> -----Original Message-----
> From: JR Richardson [mailto:jr.richardson@cox.net]
> Sent: Tuesday, March 21, 2006 4:06 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> 
> 
> > Message: 16
> > Date: Tue, 21 Mar 2006 18:51:29 -0300
> > From: "Frederic Jean" <fjean@sunnetgroup.net>
> > Subject: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> > To: <asterisk-users@lists.digium.com>
> > Message-ID: <0f7601c64d31$a05c5620$0501a8c0@snet01>
> > Content-Type: text/plain; charset="iso-8859-1"
> > 
> > Hello,
> > 
> > I am just asking this because I am note sure if the problem
> > is on my side or not, I saw some comments on SIP realtime
> > today so I was wondering, has anybody has SIP realtime working
> > with a softfone ?
> > 
> > If yes, please confirm, that would give me a light.
> > My previous message to the list is below.
> > 
> > Thanks.
> > 
> > Frederic
> 
> 
> Yes,
> 
> I have realtime working with SIP Cisco and Polycom hard 
> phones, and DIAX Softphones for IAX (pretty much same config 
> as SIP.  Sip and Iax realtime works fine for me, backing into 
> a MySQL database.  If the softphone is giving you a problem, 
> try another sofphone, there are a lot if free ones to try.  
> There is no reason why a soft phone would not work and a hard 
> phone would work, except configuration or SIP stack 
> implementation on the soft phone.
> 
> JR Richardson
> Engineering for the Masses
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-21 17:18 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

I have no idea, but that's not the point.
The point is that if you do have to, then you shouldn't lose any data.
In a production environment, the last thing you want to do is affect customers.
Given that Asterisk is supposed to be carrier-grade, I'd have thought this
was a given.

Doug.
> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> Sent: Tuesday, March 21, 2006 5:04 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> 
> 
> On Tuesday 21 March 2006 18:12, Douglas Garstang wrote:
> > I had to drop realtime with sip users. If you do a reload 
> or a restart, you
> > lose all the sip peer information (even with 
> rtcachefriends=yes). That just
> > wasn't acceptable for us.
> 
> I have to ask; with a realtime (read: database-driven) system 
> in place, why 
> did you have to reload/restart so often?  I know people who 
> never restart or 
> reload their systems and everything's coming out of the DB.
> 
> -A.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-22 09:34 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> Sent: Wednesday, March 22, 2006 8:55 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> 
> 
> On Tuesday 21 March 2006 18:12, Douglas Garstang wrote:
> > I had to drop realtime with sip users. If you do a reload 
> or a restart, you
> > lose all the sip peer information (even with 
> rtcachefriends=yes). That just
> > wasn't acceptable for us.
> 
> I just spoke to oej about this on IRC.
> 
> The only time this stuff's cleared is when you issue a reload 
> (specifically a 
> sip reload).  
> 
> Now I have to ask you this: **WHY** are you reloading on a 
> production system, 
> *especially* one using realtime?  Can you explain why this is 
> so necessary?  
> You are very keen on placing blame and saying how much of a 
> requirement 
> absolute high availability is for you, but then you talk 
> about having to 
> reload constantly.  Why are all these reloads necessary?
First thing that comes to mind, what if we decided to change a non user setting
in sip.conf?

> -A.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-23 12:54 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

Ok Andrew. Here's one for you... I just changed qualify from yes to no in
the database... a 'sip show peers' still showed Asterisk as qualifying
the users... I had to do a reload to get to accept the change to the database.

> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> Sent: Wednesday, March 22, 2006 10:46 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> 
> 
> On Wednesday 22 March 2006 11:34, Douglas Garstang wrote:
> > First thing that comes to mind, what if we decided to 
> change a non user
> > setting in sip.conf?
> 
> You're reaching.  You said you NEED to reload all the time, 
> that this is a 
> MAJOR issue, a deal breaker.  So surely you must have 
> experienced this 
> downtime to be so sensitive to it.  What did you do on your 
> PRODUCTION system 
> that required constant reloads to cause the current behavior 
> to be such a big 
> problem?
> 
> Honestly; if you're changing a non-user setting in sip.conf 
> you're going to do 
> that very, very infrequently, and you'd do it during a low 
> volume time.
> 
> You said this is a major problem.  I'm calling you on it.  
> I'm interested in 
> making Asterisk robust and highly-available too, but I'm not 
> making up 
> scenarios in order to launch complaints and verbal assaults 
> against the 
> project in order to feed my inflated ego and try to get 
> things done "my way."
> 
> If you have a specific problem, let's hear it.
> 
> -A.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-23 12:55 UTC

head link

[Asterisk-Users] Re: Fw: anybody has SIP realtime working ?

I _think_ you can do a 'sip reload' instead of a 'reload' and
keep your BLF...
But as for a server boot.... don't be crazy! According to people on this
group, that's just plain never required in the real world.

> -----Original Message-----
> From: mustardman29 [mailto:mustardman29@hotmail.com]
> Sent: Thursday, March 23, 2006 12:09 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP 
> realtime working ?
> 
> 
> Sorry for the stupid question because I don't really 
> understand the gory
> details of this as much as I'd like to.  
> 
> Do ANY of the discussions related to this particular thread 
> help address the
> Asterisk bug of BLF not working after SIP reload or server reboot?
> 
> > -----Original Message-----
> > From: Olle E Johansson [mailto:oej@edvina.net] 
> > Sent: Thursday, March 23, 2006 7:57 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP 
> > realtime working ?
> > 
> > 
> > 23 mar 2006 kl. 16.24 skrev Benny Amorsen:
> > 
> > >>>>>> "AK" == Andrew Kohlsmith
<akohlsmith-asterisk@benshaw.com>
> > >>>>>> writes:
> > >
> > > AK> If you have a specific problem, let's hear it.
> > >
> > > If I add or remove a pickupgroup or call group from a 
> phone in the 
> > > database, I need to sip reload.
> > 
> > The reason we call them dynamic peers and users is that you 
> > don't have to.
> > We look them up every time we need them and configure them 
> > with the latest data in the database.
> > 
> > Unless you turn on caching, but that's another story. SIP 
> > reload will be needed to clear them out from memory and load 
> > them again from database next time they communicate with Asterisk.
> > 
> > /O
> > 
> > 
> > ---
> > * Olle E. Johansson - oej@edvina.net
> > * Asterisk Training http://edvina.net/training/
> > 
> > 
> > 
> > 
> > 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-23 15:00 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

I also just changed the callerid column for a use in realtime. Doing a
'realtime load sippeers name 80014052' shows the OLD value for callerid.
Do I have to do a reload here? If I do, I'll lose all my sip peer
information - BAD!
> -----Original Message-----
> From: Douglas Garstang 
> Sent: Thursday, March 23, 2006 12:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: akohlsmith-asterisk@benshaw.com
> Subject: RE: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> 
> 
> Ok Andrew. Here's one for you... I just changed qualify from 
> yes to no in the database... a 'sip show peers' still showed 
> Asterisk as qualifying the users... I had to do a reload to 
> get to accept the change to the database.
> 
> 
> > -----Original Message-----
> > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> > Sent: Wednesday, March 22, 2006 10:46 AM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> > 
> > 
> > On Wednesday 22 March 2006 11:34, Douglas Garstang wrote:
> > > First thing that comes to mind, what if we decided to 
> > change a non user
> > > setting in sip.conf?
> > 
> > You're reaching.  You said you NEED to reload all the time, 
> > that this is a 
> > MAJOR issue, a deal breaker.  So surely you must have 
> > experienced this 
> > downtime to be so sensitive to it.  What did you do on your 
> > PRODUCTION system 
> > that required constant reloads to cause the current behavior 
> > to be such a big 
> > problem?
> > 
> > Honestly; if you're changing a non-user setting in sip.conf 
> > you're going to do 
> > that very, very infrequently, and you'd do it during a low 
> > volume time.
> > 
> > You said this is a major problem.  I'm calling you on it.  
> > I'm interested in 
> > making Asterisk robust and highly-available too, but I'm not 
> > making up 
> > scenarios in order to launch complaints and verbal assaults 
> > against the 
> > project in order to feed my inflated ego and try to get 
> > things done "my way."
> > 
> > If you have a specific problem, let's hear it.
> > 
> > -A.
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> > 
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-23 16:48 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

Please don't tell me what I think you are. Are you saying that to change a
configuration setting for the phone I have to remove it as a peer, and then wait
for it to re-register? Are you serious???

Doug.
> -----Original Message-----
> From: Aaron Daniel [mailto:amdtech@shsu.edu]
> Sent: Thursday, March 23, 2006 2:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> 
> 
> Maybe if I repeat myself like 18 times.
> 
> sip prune realtime <exten>
> 
> USE THIS COMMAND... it clears the settings on a single phone, 
> leaving all 
> the others untouched.  We do it all the time when we 
> reconfigure a phone.
> 
> Aaron :)
> 
> On Thu, 23 Mar 2006, Douglas Garstang wrote:
> 
> > Ok Andrew. Here's one for you... I just changed qualify 
> from yes to no in the database... a 'sip show peers' still 
> showed Asterisk as qualifying the users... I had to do a 
> reload to get to accept the change to the database.
> >
> >
> >> -----Original Message-----
> >> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> >> Sent: Wednesday, March 22, 2006 10:46 AM
> >> To: asterisk-users@lists.digium.com
> >> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime 
> working ?
> >>
> >>
> >> On Wednesday 22 March 2006 11:34, Douglas Garstang wrote:
> >>> First thing that comes to mind, what if we decided to
> >> change a non user
> >>> setting in sip.conf?
> >>
> >> You're reaching.  You said you NEED to reload all the time,
> >> that this is a
> >> MAJOR issue, a deal breaker.  So surely you must have
> >> experienced this
> >> downtime to be so sensitive to it.  What did you do on your
> >> PRODUCTION system
> >> that required constant reloads to cause the current behavior
> >> to be such a big
> >> problem?
> >>
> >> Honestly; if you're changing a non-user setting in sip.conf
> >> you're going to do
> >> that very, very infrequently, and you'd do it during a low
> >> volume time.
> >>
> >> You said this is a major problem.  I'm calling you on it.
> >> I'm interested in
> >> making Asterisk robust and highly-available too, but I'm not
> >> making up
> >> scenarios in order to launch complaints and verbal assaults
> >> against the
> >> project in order to feed my inflated ego and try to get
> >> things done "my way."
> >>
> >> If you have a specific problem, let's hear it.
> >>
> >> -A.
> >> _______________________________________________
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> Asterisk-Users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> -- 
> Aaron Daniel
> Computer Systems Technician
> Sam Houston State University
> amdtech@shsu.edu
> (936) 294-4198
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-23 18:28 UTC

head link

[Asterisk-Users] Re: Fw: anybody has SIP realtime working ?

I don't know why the situation is different, but we've been using
Polycom phones with BLF, and it's ok. I'm using Asterisk 1.2.5, and a
'reload' will clear sip subscriptions and BLF, but a 'sip
reload' does not.

Doug.

> -----Original Message-----
> From: mustardman29 [mailto:mustardman29@hotmail.com]
> Sent: Thursday, March 23, 2006 4:54 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP 
> realtime working ?
> 
> 
> Thanks BJ,
> 
> I tried your patch and it worked fine for me so thank you so 
> much for the
> effort.  It is very much appreciated.  Especially since I am 
> not capable of
> coding myself.  
> 
> Unless I can get a total solution so that it just works no matter if I
> reload or reboot then it's not really a solution for me. I 
> have to either
> not implement BLF for install something other than Asterisk.  
> 
> Telling the client that all they have to do is reboot their 
> phone everytime
> BLF stops working is not the sort of impression I want to 
> make.  Yes, it
> will probably be rare if the system is rock solid with no 
> nightly/weekly
> cron jobs to reboot at night and UPS'ed etc. but a phone 
> system feature has
> to either just work always or not be used at all IMHO.
> 
> As far as I'm concerned, BLF simply does not work because of this :(.
> 
> 
> 
> > -----Original Message-----
> > From: BJ Weschke [mailto:bweschke@gmail.com] 
> > Sent: Thursday, March 23, 2006 2:00 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP 
> > realtime working ?
> > 
> > On 3/23/06, Douglas Garstang <dgarstang@oneeighty.com> wrote:
> > > I _think_ you can do a 'sip reload' instead of a
'reload'
> > and keep your BLF...
> > > But as for a server boot.... don't be crazy! According to 
> > people on this group, that's just plain never required in the 
> > real world.
> > >
> > >
> > > > -----Original Message-----
> > > > From: mustardman29 [mailto:mustardman29@hotmail.com]
> > > > Sent: Thursday, March 23, 2006 12:09 PM
> > > > To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'
> > > > Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP
realtime
> > > > working ?
> > > >
> > > >
> > > > Sorry for the stupid question because I don't really 
> > understand the 
> > > > gory details of this as much as I'd like to.
> > > >
> > > > Do ANY of the discussions related to this particular 
> thread help 
> > > > address the Asterisk bug of BLF not working after SIP reload
or
> > > > server reboot?
> > > >
> > 
> >  Bug 6047 also allows you to do a full reload and keep your 
> > BLF. As for BLF surviving Asterisk restarts, that's not so 
> > easy because BLF's are much closer to calls themselves than 
> > they are registrations. SIP calls/transactions just don't 
> > survive an asterisk restart and there's not even a framework 
> > or design for them to start doing so at this point.
> > 
> > --
> > Bird's The Word Technologies, Inc.
> > http://www.btwtech.com/
> > 
> > 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Douglas Garstang

2006-Mar-23 23:01 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

><rant>
>*SCREAMS*
>
>Tell me, why is someone as sarcastic as you and with such a caustic
>attitude towards an OPEN SOURCE project that is maintained and fixed
>primarily by people on their OWN TIME even looking at this project.I'm not looking at this project. It's already been chosen. 
 >
>Day in and day out, you find something you don't like, and spend a week
>bitching to the asterisk-users list about how asterisk isn't worth this
>and isn't worth that.  If you don't like it, don't use it.  Go
with Cisco,
>where everything is server centric, or go grab a Nortel system, so you
>won't have to worry about registrations.Have you ever considered that maybe this is because there's plenty of things
not to like? I don't know how high your standards are, but they must not be
as high as mine. I don't seem to recollect any specific situations where I
have said asterisk isn't worth this or that. I do recollect many specific
occassions where I have asked questions because Asterisk either didn't work
the way I expected, didn't work the way the docs implied, or didn't work
in a Enterprise Class' fashion that Digium states it as. You try spending
all trying to get something apparently simple to work, and see how much it
frustrates you sometime. Your statement about me 'worrying about
registrations' tends to emphasize the fact that your standards are indeed
not very high. People using 'Enterprise Class' software expect it to
work in a 'Enterprise Class' fashion. That means that your service
doesn't go down for N number of minutes while you wait for a phone to
reregister, in order to get around some Asterisk HA limitation.
 >
>Unless I'm totally off base on this, SIP is totally phone centric, with
a
>proxy to let phones know where each other is.  Hell, you can generally
>drop a SIP phone on a desk, and just dial IP's and it wouldn't give
a
>damn.  Asterisk is just that kind of device, it doesn't have to stay in
>the phone call to know what's going on, and if you program it right, if
>you lose a phone call during conversations, you won't lose those
>conversations, just the cdr's that go with them.It does have to remain in the call path if you want DTMF services such as call
recording, which we do. Not sure where you where going with the rest of that
paragraph.
>
>Yes, you have to prune your database entries, or just wait a few minutes
>for the phone to re-register if you want information from a database
>backend to work.  That's how it works, if you don't like it, rewrite
it,
>or give some CONSTRUCTIVE criticism to the deveopers.  In here, we use
>Asterisk because we like how it works, and it works great for us.  I'm
in
>the middle of a huge rollout because we're extremely happy with the
>system.  I'm sorry you're not, and that's all I can really say
unless
>you're quite a bit more cordial about your responses and requests.Telecom customers of an 'Enterprise Class' pbx solution don't expect
to have to wait minutes, with no ability to receive calls, for updates to happen
to their account. Now, I would give constructive criticisim if I knew how it
worked. I spend all my time just trying to understand that, between bad
documentation, and downright hostile attitudes towards anything bad said about
Asterisk by the likes of yourself.  I've spent the last several hours
dicking around with Queue functionality and it's behaving not at all like
the docs say. How can I make a constructive criticism about something I
don't understane the function of? How can I make a constructive criticism
about how realtime works when it isn't properly documented anywhere? If
there where some hard and fast facts, I could counter those and constructively
say why I didn't like it, but such a thing doesn't exist so I'm
stuck with bits and pieces here and there and trying to piece it's function
together from my own observations.
>
>If I'm totally off base, ignore me, but I'm tired of the constant
remarks
>and rude comments about asterisk and it's developers.  Those of us that
>have been here for a while are here because collectively, we know what it
>takes to learn asterisk.  Hell, some of us have even helped out a little
>with the development.Which rude remarks towards developers are you specifically referring to? I
don't seem to recollect any. I have actually received a number of personal
emails from people who completely understand where I am coming from and are just
as frustrated as I, both my Asterisk's limitations (specifically HA), and
the hostile attitudes towards people who even consider questioning
Asterisk's ability to save the world.

</rant>

Aaron

On Thu, 23 Mar 2006, Douglas Garstang wrote:
> Please don't tell me what I think you are. Are you saying that to
change a configuration setting for the phone I have to remove it as a peer, and
then wait for it to re-register? Are you serious???
>
> Doug.
>
>> -----Original Message-----
>> From: Aaron Daniel [mailto:amdtech@shsu.edu]
>> Sent: Thursday, March 23, 2006 2:54 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: RE: [Asterisk-Users] Fw: anybody has SIP realtime working ?
>>
>>
>> Maybe if I repeat myself like 18 times.
>>
>> sip prune realtime <exten>
>>
>> USE THIS COMMAND... it clears the settings on a single phone,
>> leaving all
>> the others untouched.  We do it all the time when we
>> reconfigure a phone.
>>
>> Aaron :)
>>
>> On Thu, 23 Mar 2006, Douglas Garstang wrote:
>>
>>> Ok Andrew. Here's one for you... I just changed qualify
>> from yes to no in the database... a 'sip show peers' still
>> showed Asterisk as qualifying the users... I had to do a
>> reload to get to accept the change to the database.
>>>
>>>
>>>> -----Original Message-----
>>>> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
>>>> Sent: Wednesday, March 22, 2006 10:46 AM
>>>> To: asterisk-users@lists.digium.com
>>>> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime
>> working ?
>>>>
>>>>
>>>> On Wednesday 22 March 2006 11:34, Douglas Garstang wrote:
>>>>> First thing that comes to mind, what if we decided to
>>>> change a non user
>>>>> setting in sip.conf?
>>>>
>>>> You're reaching.  You said you NEED to reload all the time,
>>>> that this is a
>>>> MAJOR issue, a deal breaker.  So surely you must have
>>>> experienced this
>>>> downtime to be so sensitive to it.  What did you do on your
>>>> PRODUCTION system
>>>> that required constant reloads to cause the current behavior
>>>> to be such a big
>>>> problem?
>>>>
>>>> Honestly; if you're changing a non-user setting in sip.conf
>>>> you're going to do
>>>> that very, very infrequently, and you'd do it during a low
>>>> volume time.
>>>>
>>>> You said this is a major problem.  I'm calling you on it.
>>>> I'm interested in
>>>> making Asterisk robust and highly-available too, but I'm
not
>>>> making up
>>>> scenarios in order to launch complaints and verbal assaults
>>>> against the
>>>> project in order to feed my inflated ego and try to get
>>>> things done "my way."
>>>>
>>>> If you have a specific problem, let's hear it.
>>>>
>>>> -A.
>>>> _______________________________________________
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation provided by Easynews.com --
>>>
>>> Asterisk-Users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> Aaron Daniel
>> Computer Systems Technician
>> Sam Houston State University
>> amdtech@shsu.edu
>> (936) 294-4198
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
amdtech@shsu.edu
(936) 294-4198
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Douglas Garstang

2006-Mar-23 23:04 UTC

head link

[Asterisk-Users] Fw: anybody has SIP realtime working ?

If you referring to me Aaron, I don't recollect every using the word
'stupid'. I have said many times that the way certain things are done
for an Enterprise Class piece of software are unacceptable, or maybe reacted in
incredulation that it would even be considered being done in a certain way.
 
It's quite obvious that any negative statements about Asterisk are not taken
too well.

	-----Original Message----- 
	From: Aaron Daniel [mailto:amdtech@shsu.edu] 
	Sent: Thu 3/23/2006 10:38 PM 
	To: Asterisk Users Mailing List - Non-Commercial Discussion 
	Cc: 
	Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
	
	

	> Yipes, Aaron.  The point of a troll is to elicit exactly the behavior you
	> just exhibited.
	>
	> If everyone were to adopt a really simple protocol: "Ask nicely,
Doug, or
	> we'll all remain silent," he would either start behaving or go
away.
	>
	> I'm sure all the tumult he elicits is a vast ego boost for him.  Look
how one
	> little guy with a tart mouth is able to keep an entire mailing list roiled
up
	> for weeks at a time.
	>
	> Next one of these I write goes off-list.  It is, however as you have
	> discovered, very very difficult to keep silent in the face of such boorish
	> behavior.
	>
	> B.
	
	Yeah, I'm rather embarrassed that I even wrote that, it's just
	frustrating.  We're here to help people, but when they completely
	disregard everything anyone says about the project as stupid or other
	various notions, it's kinda grating.  We've all had to go through a lot
of
	what he's going through, especially with a project with as high a learning
	curve as asterisk, so I know where he's coming from, but when you're
	asking for help, ask for help, don't dog the project.  That's all I was
	really trying to get at :)
	
	Sorry if I upset anyone, just been one of those days.
	
	--
	Aaron Daniel
	Computer Systems Technician
	Sam Houston State University
	amdtech@shsu.edu
	(936) 294-4198
	_______________________________________________
	--Bandwidth and Colocation provided by Easynews.com --
	
	Asterisk-Users mailing list
	To UNSUBSCRIBE or update options visit:
	   http://lists.digium.com/mailman/listinfo/asterisk-users

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