[Asterisk-Users] Re: 12SP registration
A further question... How would you support multiple 12SP's on an asterisk install if only one can be hardcoded into the source/binary? Is this a mini programming project to add the magic number to the /etc/asterisk/skinny.conf, and then have chan_skinny.c search the config file? Ta muchly. Roger De Salis -- Date: Tue, 24 Feb 2004 09:57:57 - Subject: Re: [Asterisk-Users] 12SP From: Robert Boardman <[EMAIL PROTECTED]> Hi Cullen You need to change the hard coded firmware in chan_skinny.c line 71 ish > to the numbers that show up when your phone boots, re compile and all > will be fine Robb Cullen Simpson <[EMAIL PROTECTED]> said: I am trying to get a Cisco 12SP phone to work with *. I do not have call manager. When start * and turn skinny debugging on I get this on the console: -- -- Starting Skinny session from 192.168.1.202 Recieved AlarmMessage Device SEP0010EB003E03 is attempting to register -- Device 'ipme' successfuly registered Requesting capabilities Version Request Received CapabilitiesRes Feb 23 16:29:29 WARNING[794722]: chan_skinny.c:2275 get_input: Skinny Client sent less data than expected. Feb 23 16:29:29 NOTICE[794722]: chan_skinny.c:2333 skinny_session: Skinny Session returned: Success --- The phone indicates that it is programming. The IP address of the phone is correct in the logs. Here is a snippet from my skinny.conf file: --- [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 192.168.1.11 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 ; allow = all ; disallow = ; Typical config for 12SP+ [ipme] device=SEP0010EB003E03 version=P002G204; Thanks critch context=outbound-analog line => 120 ; Dial(Skinny/[EMAIL PROTECTED]) - Any ideas? -- Cullen Simpson [EMAIL PROTECTED] ___________ -- \_Roger De Salis rdesalis at fx dot net dot nz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outdial digits - non TDM trunk
I have successfully built and made asterisk talk SIP extension to SIP extension, read all the docs, and about 1000 emails from the archive. The trunk side of Asterisk, from the docs perspective, is a smidgin TDM-centric, Analogue, T1, zaptel.conf etc. Asterisk cares not about the externally presented digits as the telco KNOWS which time-slot or analogue line the call came from I live in an hybrid H323 IP trunk world, and when dialing out, I see the dialled digits in the (non-asterisk) H323 debug traces along the with the 4 digit extension code as the src digits. How do you prepend the remainder of the number to the src digits? extension = 2201external view 460 899 2201 Or do I need to make up two extension values, and only use one internally, and the other externally.. Ta muchly -- \_Roger De Salis rdesalis at fx dot net dot nz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk H323 Trunk
During debugging of H323 trunk side (using Jeremy Macnamara's H323
driver in ~/channels/h323) a couple of things come don't quite work
as advertised...
1/ the following line in extensions.conf explicitly sets the
outgoing caller ID (required in my case for downstream GK
processing..)
exten => _1NX.,1,SetCallerID,6400047602100
exten => _1NX.,2,Dial,H323/${EXTEN:1}
what comes out in the h323 datastream and logs is:-
dialled digits
{
{src digits}
{ "6400047602100)" }
...
note the unbalanced closing backet. We tried changing the
number length, and doing called ID with different functions,
but it looks like a bug.
A fairly detailed squiz around the digium site did not point
where to file a bug to, so I apologise for polluting this list..
=
When the h323 channel driver registers..
vipe50#sho gatek endp
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags
--- - --- - - -
202.37.19.101720 202.37.19.101024 zone1 TERM
E164-ID: 6400047602201
202.37.19.111720 202.37.19.111719 zone1 TERM
E164-ID: 6400047602999
;
; Asterisk, strongly preferred as a VOIP-GW, but registers as a TERM.
;
202.37.19.121720 202.37.19.1232829 zone1 TERM
H323-ID: fxchange
E164-ID: 6400047602100
;
; Cisco Call Manager, registering as a VOIP GW, with H323 Trunk
;
202.37.83.101720 202.37.83.101710 zone1 VOIP-GW
H323-ID: 202.37.83.10
203.79.85.252 1720 203.79.85.252 1719 zone1 TERM
E164-ID: 6400047601020
I had a look through the source, no comments stood out, any know a way
to get * registered as a VOIP-GW, rather than a TERM? Played with all
the obvious things in h323.conf
Many Thanks for reading this far...
Rgds Roger De Salis
--
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[Asterisk-Users] Cisco 7920 phone
John Todd wrote > Cisco has an 802.11 phone called the 7920, which is apparently > shipping now. It is very expensive (>$550 USD) and only runs SCCP at > the moment, which is Cisco's proprietary VoIP protocol. However, if > it falls in line with some of Cisco's other high-end VoIP equipment, > that means it should have a trailing-edge SIP image running by > December. Despite Cisco's frequent stupidity when it comes to > marketing decisions, they make some pretty nice gear and so I'm > holding out for the 7920. > > JT I have two on my desk. And very nice they are too. Found the hidden menus, from whence you can crank the power from standard 20mW to 100mW, for those "just gotta talk moments..." Battery lasts about 1 hour talk time, 8 hours standby... (but that was on the 20mW setting)... We are trying them out around Wellington, on the citywide Wifi network. They work very very well. Interesting menu options implying mechanisms to take the 11 channels of WiFI, and dedicate 1-3 for voice, and turn the rest over to data. There were some rumours that they only work on Cisco Aironet base stations They work fine on DLink, Kamaguza, and Uncle Tom Cobblies base stations... SCCP only, and don't hold your breath -- \_Roger De Salis rdesalis at fx dot net dot nz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Structured release, Maillists
From the thread. Subject: Re: [Asterisk-Users] IAX <> IAX trunking... DP cache? Date: 20 Aug 2003 11:29:59 -0600 From: Steve Meyers <[EMAIL PROTECTED]> Brian West wrote: I would use the latest CVS for one. And try again. Unfortunately, I've tried numerous times to get a current CVS trunk snapshot to talk to *anything*, to no avail. Even getting my Grandstream phones to register with it was an apparent excersize in futility. Dropping back to 0.4.0 *immediately* worked with the same configs. I'll give it a go again with today's snapshot and see if I can get *anything* to work again. Is there any hope for a 0.5.0 release on the horizon? I would also like to see a more structured release program. It's kind of scary to tell people that they should "just use the latest CVS code". I agree with this. Any chance of a "version.h" in the top level directory. It would add to confidence level a fraction... == How hard would it be to drop all the HTML and =20 messages out of the maillist digest..... This would improve the s/n ratio... Many Thanks Roger De Salis Steve -- \_Roger De Salis rdesalis at fx dot net dot nz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ATAs
John Todd wrote > For those of you wanting to salvage your Cisco ATA-186 after > inadvertent locking, or after recovering your devices from a vendor > who has locked them, here is a rainy-day project for you: > somedoc.pdf. Immediately strides to ATA, rips off cover... woohoo, EEPROM is socketed well maybe I'll just copy the contents of a working ATA into the programmer, and reflash the locked one, taking care to change the MAC address and serial number.. > Please aim your negative karma at Cisco for creating a piece of > hardware that can be rendered useless with software. This is against > all previous ideology of Cisco, and is a disturbing trend. > > JT Amen to that... -- \_Roger De Salis rdesalis at fx dot net dot nz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1273 - 10 msgs
> Message: 5 > Date: Fri, 12 Sep 2003 13:24:13 -0400 (EDT) > From: "David C. Troy" <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > All, > > I know you can use, say, a 2620 w/2 port FXO card as > a SIP gateway. Clearly you can use the 5300, 5800, and > MGX8850 too. > > Does anyone know which cards, if any, exist for a 7206VXR > to act in a similar capacity, either as a T1/PRI, DS3, or PA-VXA-1TE1-24+ 1 Port T1/E1 Digital Voice Port Adapter with 24 Channels B $7,500 which in practical terms, means the DSP farm..., but wait, you need licenses, and an IOS version that contains the features you need (is this possible?), and .. you get the idea... > POTS FXO/FXS? Nope. Kinda like saying - I need a 7206VXR to support 3 home internet users. would be very cool... (I bet there is someone on this list that comes close...) (JT - speak up...) > What other Cisco routers can act as SIP gateways today? Are Cisco's green? Anything with an FXO port and a suitable IOS load... (The correct answer to this question cannot be reasonable expressed on a mail list, it is more like a SQL query, except Cisco does not make a Sales SQL database, preferring instead that you wade through pages and pages of mindless marketing speak on www.cisc.com... > Thanks, > Dave -- \_Roger De Salis rdesalis at fx dot net dot nz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
