[Asterisk-Users] SPA941 et al LED indications
Hi all. The SPA941 and friends have pretty multicoloured LEDs, but there doesn't appear to be any support for SUBSCRIBE/NOTIFY as * as implemented for extension hinting. Has anyone managed to get the phone to support this? Thanks! -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ --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
Re: [Asterisk-Users] Uniden UIP200 Opinions
On Fri, 2005-08-05 at 13:30 -0400, Jim Feniello wrote: I've read through the archives, and wanted to get an updated opinion on the Uniden UIP200 phone. Seems like there were a lot of opinions that it was a good phone, but there were a few items that people were waiting for firmware updates for, but that was in 2004. We've deployed about 50 here. They work, mostly. Hold works (* does MOH when on hold), transfers kinda work (using the XFER button, the phone does seem to occasionally get confused afterwards tho, but * does MOH), DND and Forwarding both work. But, I would fall short of recommending them. Would really like to see the transfer problems resolved. That and the documentation is sub-par. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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] Any one using the new Digium echo cancellation cards
On Wed, 2005-08-17 at 08:56 -0500, Alan Bunch wrote: THe wiki doesn't seem to have any user reports. If your using them, how are the working, better, worse about the same. Also what hardware seems to be stable with them installed. I'd also be interested if the module is available as an upgrade to existing quad boards. It looks rather like the echo canceller is a daughter board, connected roughly where there's a connector on existing quad boards... Have a rather nastry PRI echo problem which so far no fiddling with settings (tx, rx, taps) has helped with. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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] Unknown signalling 896?
On Wed, 2005-03-16 at 04:18 -0600, Eric Wieling wrote: David Zanetti wrote: But * won't bring up chan_zap at all: ERROR[2215]: Signalling requested is PRI Signalling but line is in Unknown signalling 896 signalling ERROR[2215]: Unable to register channel '1-30' WARNING[2215]: chan_zap.so: load_module failed, returning -1 WARNING[2215]: Loading module chan_zap.so failed! Ideas? I'm sure it's something simple I've missed. :) Config fragments follow: Notice the [channels] lines at the top of the zaptel.conf.sample? You need it. Ah, I just didn't paste that into the email from the fragments I chose. I didn't want to just post the whole file. It's there. * tries to configure them, just fails. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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
Resolved: Re: [Asterisk-Users] Unknown signalling 896?
On Thu, 2005-03-17 at 09:40 +1300, David Zanetti wrote: ERROR[2215]: Signalling requested is PRI Signalling but line is in Unknown signalling 896 signalling ERROR[2215]: Unable to register channel '1-30' WARNING[2215]: chan_zap.so: load_module failed, returning -1 WARNING[2215]: Loading module chan_zap.so failed! Helps if I remember it doesn't renumber the channels, you still carry over the 1-15,17-31 thing from /etc/zaptel.conf to /etc/asterisk/zapata.conf. (E1s here have their D channel as 16, the rest are B channels..) Opps. :) -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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] Unknown signalling 896?
I've been beating my head a bit against the 1.0.6 Debian builds of Asterisk, using an E100P (E1, single span) board. In machines I've built in the past (back in 1.0.0 days), config I'm using and that card and 1.0.0 driver combo worked fine. ztcfg reports no problems: SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels configured. And zttool sees the card, and reports it in the state I expect (there's no real E1 attached to it, so blue/red alarms..) But * won't bring up chan_zap at all: ERROR[2215]: Signalling requested is PRI Signalling but line is in Unknown signalling 896 signalling ERROR[2215]: Unable to register channel '1-30' WARNING[2215]: chan_zap.so: load_module failed, returning -1 WARNING[2215]: Loading module chan_zap.so failed! Ideas? I'm sure it's something simple I've missed. :) Config fragments follow: ==/etc/zaptel.conf== span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=nz defaultzone=nz ==end== ==/etc/asterisk/zapata.conf== context = default switchtype = euroisdn priindication = outofband group = 2 signalling = pri_cpe channel = 1-30 ==end== Thanks. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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] Dialogic D/300JCT-E1 support
Hi all.. Both voip-info.org, Digium's own site, and asterisk.org are a bit light on the details on support of Dialogic's single E1 span. Is the single E1 span supported, what channel does it get configured as, and does anyone have a guide or some experience in getting it function in *? Mailing list searches suggest I need to purchase a channel driver from Digium, but no obvious link to such a driver seems to be on the website.. Hopefully it doesn't require binary-only kernel code, it could be a bit painfull.. :| Thanks :) -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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
Re: [Asterisk-Users] Dialogic D/300JCT-E1 support
On Fri, 2004-10-15 at 13:16, Steven Critchfield wrote: On Fri, 2004-10-15 at 10:47 +1300, David Zanetti wrote: Mailing list searches suggest I need to purchase a channel driver from Digium, but no obvious link to such a driver seems to be on the website.. Hopefully it doesn't require binary-only kernel code, it could be a bit painfull.. :| I think you will be in a binary only kernel module just to use the dialogic card under linux. You have to contact Digium by phone to see about purchaseing the channel driver. I don't think it is binary only, but wouldn't surprise me. Right, I shall give them a call then. You would do well to ebay the card if you don't otherwise need it and then buy a Digium card. Alas, the Digium cards (which I have used and are great) do not carry a TelePermit for New Zealand - it's illegal to connect them to the PSTN here. Until the Digium cards are made legal, I have to look at alternatives :( -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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] No call progress from * to E1
Setup: System Phones -- PBX --E1-- * -- SIP Phones Calls work in both directions. However, ringing feedback to caller only works in the SIP-System direction. System callers to a SIP endpoint get silence, until call is picked up or dropped into something else. I've tried both Dial() with the r option, and Ringing() before Dial(), and neither works. With PRI debugging enabled, I can see Asterisk never passes a call progress message back to the PBX. Bug? Option I've missed? Thanks. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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
Re: [Asterisk-Users] QSIG against a Nortel/Meridian PBX
On Fri, 2004-09-10 at 04:50, [EMAIL PROTECTED] wrote: QSIG passes callername and other variables by a mechanism that asterisk cannot interpret at the moment. It sends them either in a information element in the setup message for the call or in an additional facility message after the fact. Right now asterisk cannot interpret those facility messages to get the data out of them, so that's probably the reason why you are getting garbled data. If anyone has any info on interpreting the FACILTIY message/IE we could probably get something put together to interpret it. Parsing would be a nice to have, but I'd settle for FACILITY not nuking the callerid entirely. If FACILITY messages are unparsable, shouldn't they be dropped by *? (I can ask the PBX provider for documentation of the FACILITY they are sending, but it seems like a bug to me that a long message causes the _number_ in a seperate IE to be dropped.) Thanks. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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] QSIG against a Nortel/Meridian PBX
[Reposting, as was bounced for non-member, sorry if this is a dupe]
Arrangement:
{ PSTN }--E1--[PBX]--E1--[*]--LAN--[SIP phones]
\__[PBX system phones]
Normal calls between PBX system phones and SIP phones work, in both
directions. The call logs look like (ignore the no answer, it did ring):
,7436,7902,e1-in,^B^B^A 7436,Zap/7-1,SIP/7902-5fb7,
Dial,SIP/7902|20|t,2004-09-08 07:40:07,,
2004-09-08 07:40:08,1,0,NO ANSWER,DOCUMENTATION
There's a odd bit of data in the name portion of the callerid. I believe
that's being provided over QSIG from the PBX (and the people who
maintain the PBX confirmed QSIG is being sent..)
If a forced redirection is set on the PBX to a SIP extension, the PBX
sends quite a long QSIG message, and we end up with this in the call
logs (phone never rings):
,,7902,e1-in,^B^B^G^A^B^A^U0U^B^A^A
^A^A80^A^AA1^MA0^KA5
^A^D^R^D7441A2^MA0^KA5
^A^D^R^D7441A3^N80^LAaron KrivanA4^N80^LAaron Kriv,
Zap/31-1,SIP/7902-f8de,Dial,SIP/7902|20|t,
2004-09-08 12:00:02,,2004-09-08 12:00:14,12,0,
NO ANSWER,DOCUMENTATION
It passes all this over SIP to the phone, but the phone is dumb to it.
The From field contains that name string as it is above (as binary,
instead of ^A etc..), but does not contain a number section.
I had thought to just use SetCallerID to supress the name portion, but
there's no number portion in ${CALLERNUM}. It's like the name part is so
long it's overflowing the buffer for the caller id.
Has anyone else successfully got such an arrangement to work, or is
there any plans to make the QSIG messages more parsable (maybe exposed
as variables in the dialplan)? Or at least not have the name overflow
the number? :)
Thanks.
--
David Zanetti [EMAIL PROTECTED]
Team Leader, Systems Administration
+64-4-8032233 +64-21-402260
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