Saturday, May 24, 2008

Re: [asterisk-users] [asterisk-biz] New York Asterisk Users

I aam in South Jersey would love to participate

On 5/24/08, | dave cantera | <david.cantera@iacnet.net> wrote:
> dean,
> I am an active member of AUG NYC... you can email me off list for any info
> you need.
>
> also, I am preparing to start a south jersey * UG. the phila group is
> waning...
>
> thanks,
> daveC
>
> Dean Collins wrote:
>>
>> This is an email to all New York based Asterisk users.
>>
>>
>>
>> For some time it's been bugging me that we don't have a local contact
>> point/user community. If you are involved in Asterisk and in NY/NJ shoot
>> me an email, I'm going to try and revitalize either meetup.com or some
>> other shared environment for Asterisk users in NY.
>>
>>
>>
>> Shoot me an email and once I get an idea of how many Asterisk users there
>> are in NY we'll work out what to do from there.
>>
>>
>>
>>
>>
>>
>>
>> Cheers,
>> Dean
>>
>>
>>
>> ________________________________
>> _______________________________________________
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>>
>> asterisk-biz mailing list
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>>

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>>
>> ________________________________
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>> Checked by AVG.
>> Version: 7.5.524 / Virus Database: 269.23.21 - Release Date: 05/19/2008
>> 12:00 AM
>>
>
>
> --
> My wife's sister is in California.
> I should buy her a Videophone2008!
>
> Truly, The Next Best Thing to Being There!
> --
>
> WorldWideVideoPhones.com
> 856.380.0894
>
>
>

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Re: [asterisk-users] Error after upgrading from 1.2.18 to 1.4.20

Hi I better answer my own post.

I went to the code and the issue is in q931.c

/* wait for a RELEASE so that sufficient time has passed
for the inband audio to be heard */

if (c->progressmask & PRI_PROG_INBAND_AVAILABLE)
break;

Changing this line to a comment makes the 1.4 work exactly as 1.2 for
this issue.
I think that this line should only be executed on 'outbound' pri calls,
not on inbound.

Freddi


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[asterisk-users] Error after upgrading from 1.2.18 to 1.4.20

scenario is incoming calls on ZAP (TE410p euro isdn) to SIP (or any
other channel) and call is answered.

When I hangup on the ISDN side on the 1.2 then the SIP hangs up to
immidiatly so everything is fine (se short pri debug below).

When I do the same on 1.4.20 then it take more than 30 seconds to
disconnect when the ISDN calling party hangs up

I have tried to use the use the 1.4 zaptel with the 1.2 installation,
that works perfect to so to me it looks like a libpri or chan_zap problem.
I do hope though that some has a hint to where I scre... up.

Freddi


OK trace from 1.2

< Protocol Discriminator: Q.931 (8) len=13
< Call Ref: len= 2 (reference 7603/0x1DB3) (Originator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
< Ext: 1 Cause: Normal Clearing (16), class = Normal
Event (1) ]
< [1e 02 82 88]
< Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0)
0: 0 Location: Public network serving the local user (2)
< Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (cs0, Cause)
-- Processing IE 30 (cs0, Progress Indicator)
-- Channel 0/14, span 1 got hangup request
== Spawn extension (Pstn-incoming-fwd, 77348855, 2) exited non-zero on
'Zap/14-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
> Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 7603/0x1DB3) (Terminator)
> Message type: RELEASE (77)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
> Ext: 1 Cause: Normal Clearing (16), class = Normal
Event (1) ]
-- Hungup 'Zap/14-1'
< Protocol Discriminator: Q.931 (8) len=5
< Call Ref: len= 2 (reference 7603/0x1DB3) (Originator)
< Message type: RELEASE COMPLETE (90)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
fonet2*CLI

BAD TRACE from 1.4 below:

fonet3*CLI>
< Protocol Discriminator: Q.931 (8) len=13
< Call Ref: len= 2 (reference 7582/0x1D9E) (Originator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: User (0)
< Ext: 1 Cause: Normal Clearing (16), class = Normal
Event (1) ]
< [1e 02 82 88]
< Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Public network serving the local user (2)
< Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (cs0, Cause)
-- Processing IE 30 (cs0, Progress Indicator)
q931.c:3779 q931_receive: call 7582 on channel 13 enters state 12
(Disconnect Indication)
fonet3*CLI>

~ 30 SECOND DELAY HERE.

fonet3*CLI>
fonet3*CLI>
< Protocol Discriminator: Q.931 (8) len=9
< Call Ref: len= 2 (reference 7582/0x1D9E) (Originator)
< Message type: RELEASE (77)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: User (0)
< Ext: 1 Cause: Normal Clearing (16), class = Normal
Event (1) ]
-- Processing IE 8 (cs0, Cause)
q931.c:3754 q931_receive: call 7582 on channel 13 enters state 0 (Null)
-- Channel 0/13, span 2 got hangup, cause 16
== Spawn extension (Pstn-incoming-fwd, 77348855, 2) exited non-zero on
'Zap/44-1'
-- Executing [h@Pstn-incoming-fwd:1] Hangup("Zap/44-1", "") in new stack
== Spawn extension (Pstn-incoming-fwd, h, 1) exited non-zero on 'Zap/44-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
Request
> Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 7582/0x1D9E) (Terminator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: Private network serving the local user (1)
> Ext: 1 Cause: Normal Clearing (16), class = Normal
Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/44-1'fonet3*CLI>


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Re: [asterisk-users] Manual Wardialer



On Sat, Apr 26, 2008 at 7:13 PM, Brian J. Murrell <brian@interlinx.bc.ca> wrote:
On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
> Does anyone have a script for manual wardialer for asterisk? not sure
>  if "wardialer" is the correct term but basically I want to call X
>  number say 555-0000 through 555-0050 and be able to listen to each
>  call and when I hang up or press a key it will dial the next number
>  for me. I guess sort of like "scanning" an exchange but I want to be
>  on the line and if possible complete / talk on certain calls.

Legal issues aside, have you tried this?

http://www.softwink.com/iwar/

Thanks,
Matt

Re: [asterisk-users] Manual Wardialer

I would say anyone in the USA that is on the donotcall.gov list should
also be excluded -- unless you either qualify as an exemption or like
paying $10k for each violation.

Brian J. Murrell wrote:
> On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
>> Does anyone have a script for manual wardialer for asterisk? not sure
>> if "wardialer" is the correct term but basically I want to call X
>> number say 555-0000 through 555-0050 and be able to listen to each
>> call and when I hang up or press a key it will dial the next number
>> for me. I guess sort of like "scanning" an exchange but I want to be
>> on the line and if possible complete / talk on certain calls.
>
> Hrm. I wonder what you could possibly want that for? Do you mind if
> include my area code and exchange in your script as an exception to your
> "scanning"? Anyone else care to be on that exception list? Actually,
> it might be more efficient for me to ask who cares to be on the list to
> be scanned. Anyone? Anyone at all? ;-)


--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.

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Re: [asterisk-users] [asterisk-biz] New York Asterisk Users

dean,
I am an active member of AUG NYC... you can email me off list for any info you need.

also, I am preparing to start a south jersey * UG.  the phila group is waning...

thanks,
daveC

Dean Collins wrote:

This is an email to all New York based Asterisk users.

 

For some time it’s been bugging me that we don’t have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I’m going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY.

 

Shoot me an email and once I get an idea of how many Asterisk users there are in NY we’ll work out what to do from there.

 

 



Cheers,
Dean

 

 
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No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.21 - Release Date: 05/19/2008 12:00 AM

--  My wife's sister is in California.   I should buy her a Videophone2008!  Truly, The Next Best Thing to Being There! --  WorldWideVideoPhones.com 856.380.0894   

Re: [asterisk-users] Asterisk Database Handling

Tilghman and Jay,

Thanks for the licensing advice. If anyone is interested in replicate,
I'm now ready to distribute it under the GPL.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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