date:20050316

Re: [Asterisk-Users] VoipJet Terms of Service

2005-03-16 Thread Michael Puchol


Jean-Michel Hiver wrote:
But then again come to them with a few million monthly minutes under 
your belt and I'm sure they'll change the TOS for you...
Maybe not, as the ToS also state:
The customer agrees to purchase VoipJet termination in small amounts
What does this mean? We have to start with 5-minute calls max, then
slowly increase absolute timeout? How small is small?
How about this one:
VOIPJET DOES NOT SUGGEST, AND VEHEMENTLY DENIES, ANY CLAIM THAT ITS
VOIP SERVICES HAVE A LEVEL OF QUALITY OR RELIABILITY ANYWHERE NEAR THAT
OF THE REGULAR PHONE SYSTEM
BWAHAHAHAHAHAHA this is like saying our system sucks and we know 
it. How can they seriously expect anyone that reads this ToS to want to 
sign up with them? It would have been simpler to simply state the usual 
we cannot guarantee 100% reliability or availability of our service, 
which depends on third parties over which we have no control or 
something along these lines.

Nice laugh, best regards,
Mike
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Agent groups broken in queues? (do not follow strategy)

2005-03-16 Thread Bill Petrisko

I attempted setting up a queue with agents that log in, and
get called with incoming calls:

Agents log in using:
  exten = *88,1,AgentCallbackLogin(${CALLERIDNUM}|[EMAIL PROTECTED])

Calls get into the queue with:
  exten = 6029995654,1,Queue(test-noc|t|||60)

queues.conf:
  [test-noc]
  strategy = rrmemory
  context = test-sip
  timeout = 10
  retry = 4
  member = Agent/@2

agents.conf:
  [agents]
  ackcall=no
  group=2
  agent = 6029995670,,Joe Bob
  agent = 6029995671,,Billy Dude

Agents can log in fine, but all calls end up at the phone of 
the first agent to log in.  Always.  No matter how many people
are logged in as agents.

After beating myself up on this all night (and then finally getting
into the wiki after it was unavailable) - I come up with the following
note in the queue config section:

   http://www.voip-info.org/wiki-Asterisk+config+queues.conf
   
   If you include groups in your queue definition the calls get 
   routed in the order of the group regardless of the specified 
   strategy. So I just have a member= line for each agent. 

Is this really true?

If so, what is the point of having the ability for people to log
in/out, if it completely ignores the strategy for call distribution?

Anyone know of any plans of fixing or improving this behavior, to 
make logged in agents consistent with 'permanent' agents?

Or is there something i'm missing, and there really is a way to have
a dynamic agent follow the call strategy?

Also, in a ringall strategy, is there a maximum number of destinations?
(Either known in the code, or if someone has tested to an unusually
high number?)

Any information appreciated!

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

Re: [Asterisk-Users] OT: Best DB

2005-03-16 Thread tim panton

On 15 Mar 2005, at 23:52, Giudice, Salvatore wrote:
we were able to handle a peak of 700k inserts per hour. MySQL gave us
very few problems and probably had a cumulative downtime of
approximately 4 days per year until the project was decommissioned. 
When
y
That's more than 1% downtime, not even two nines .
What's your downtime worth per day?
http://www.westhawk.co.uk/
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Realtime does not work yet, ... bug

2005-03-16 Thread Martijn van Oosterhout

On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote:
 Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14, dst=1259
 Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL: UPDATE 
 sip_buddies SET name = '621' WHERE allow = 'g729'
 Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Everything is fine.
 Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Updated 0 rows on table: 
 sip_buddies

*ALARM* Where is that query fron? It's totally wrong! It just changed the
name of anyone who is allowed to use g729.

Looks like Realtime is not quite there yet for production...

Have a nice day,
-- 
Martijn van Oosterhout
Ecomtel Pty Ltd
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary?

2005-03-16 Thread Kamran Ahmad

hello

i try to call from sip phone on asteris to open phone
on GnuGK.
can any one tell me why it is saying

chan_oh323.c:2501 ast_oh323_new: Internal channel
initialization failed. Bad binary?
Mar 16 13:28:46 WARNING[5963]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
structure 4.
Mar 16 13:28:46 NOTICE[5963]: app_dial.c:749
dial_exec: Unable to create channel of type 'OH323'
We're at 192.168.0.203 port 17456


Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From:sip:[EMAIL PROTECTED];
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 21 INVITE
Contact: sip:[EMAIL PROTECTED]
Max-Forwards: 5
User-Agent:SKYPHONE/1.03
Subject: hello
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS,
REFER,SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:180
Proxy-Authorization: Digest
username=2000,realm=asterisk,nonce=6ebe9c68,uri=sip:192.168.0.203,response=7027ef8069a0ef7a5f8089fda2fc0e87
  
  
  
  
v=0
o=sibtay 2890844 842807 IN IP4 192.168.0.153
s=SDP Seminar
c=IN IP4 192.168.0.153
t=0 0
m=audio 13064 RTP/AVP 0 101
a=rtpmap:101 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:96 0-11,16
  
  
  
  
15 headers, 11 lines
Using latest request as basis request
Sending to 192.168.0.153 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.153:13064
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer
- audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4
(ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1
(g723), combined - 0x1 (g723)
Found user '2000'
Looking for 321 in default
list_route: hop: sip:[EMAIL PROTECTED]
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.153;branch=z9hG4bK2038176231
From: sip:[EMAIL PROTECTED];
To: sip:[EMAIL PROTECTED];tag=as61b12c41
Call-ID: [EMAIL PROTECTED]
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
  
  
  
  
 to 192.168.0.153:5060
Mar 16 13:28:34 ERROR[5963]: chan_oh323.c:2501
ast_oh323_new: Internal channel initialization failed.
Bad binary?
Mar 16 13:28:34 WARNING[5963]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
structure 3.
Mar 16 13:28:34 NOTICE[5963]: app_dial.c:749
dial_exec: Unable to create channel of type 'OH323'
  
  
*CLI
*CLI
  
  
Sip read:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: sip:[EMAIL PROTECTED]
To: sip:172.16.0.32
Call-ID: [EMAIL PROTECTED]
CSeq: 22 INFO
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/dtmf-relay
Content-Length: 26
  
  
  
  
  
  
Signal= 8
Duration= 160
  
  
9 headers, 4 lines
Receiving DTMF!
* DTMF received: '8'
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.153
From: sip:[EMAIL PROTECTED]
To: sip:172.16.0.32;tag=as61b12c41
Call-ID: [EMAIL PROTECTED]
CSeq: 22 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
  
  
  
  
 to 192.168.0.153:5060
  
  
  
  
Sip read:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.153
From: sip:[EMAIL PROTECTED]
To: sip:172.16.0.32
Call-ID: [EMAIL PROTECTED]
CSeq: 23 INFO
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/dtmf-relay
Content-Length: 26

Re: [Asterisk-Users] Realtime does not work yet, ... bug

2005-03-16 Thread Ronald Wiplinger

Martijn van Oosterhout wrote:
On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote:

Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14, dst=1259
Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL: UPDATE
sip_buddies SET name = '621' WHERE allow = 'g729'
Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Everything is fine.
Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Updated 0 rows on table:
sip_buddies

*ALARM* Where is that query fron? It's totally wrong! It just changed the
name of anyone who is allowed to use g729.
Looks like Realtime is not quite there yet for production...
Have a nice day,

vpbx*CLI help realtime update
Usage: realtime update family colmatch value
Update a single variable using the RealTime driver.
(I guess Matthew typed to fast to give me the hint, ...)
Thanks for pointing it out, it should therefore be:
vpbx*CLI realtime update sippeers name 621 allow ulaw,alaw
However, it shows:
vpbx*CLI realtime load sippeers name 621
Column Name Column Value

id 1
name 621
callerid Demo,621
canreinvite yes
context inhouse
dtmfmode rfc2833
host dynamic
mailbox [EMAIL PROTECTED]
nat yes
pickupgroup 1
qualify 999
secret Morgen621
type friend
username 621
allow ulaw,alaw
disallow all
regseconds 0
cancallforward yes

... so it has updated it!!!
... but:
vpbx*CLI sip show peer 621
vpbx*CLI
* Name : 621
Secret : Set
MD5Secret: Not set
Context : inhouse
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup : 1, 33
Mailbox : [EMAIL PROTECTED]
LastMsgsSent : 2
Inc. limit : 0
Outg. limit : 0
Dynamic : Yes
Callerid : Demo 621
Expire : 140506
Expiry : 900
Insecure : no
Nat : Always
ACL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr-IP : 192.168.250.114 Port 5060
Defaddr-IP : 0.0.0.0 Port 5060
Def. Username: 621
Codecs : 0x0 (nothing)
Codec Order : (none)
Status : OK (5 ms)
Useragent: Grandstream BT100 1.0.5.18
Full Contact : sip:[EMAIL PROTECTED]:64655
so, two different ways of the same query gives two different results!
I agree with you, ... there is something not completely working ;-)
bye
Ronald
--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208
- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org
PS: Spam prevention!
Our system is protected with a spam prevention program.
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please.
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.

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

1 2 3 >

1 - 100 of 253 matches

Mail list logo