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...n port on the module. So 54 keys max. -----Original Message----- From: Rob Lith [mailto:rob@connection-telecom.com] Sent: Wednesday, February 08, 2006 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Handset phone to replace Flash Operator Panel Garth The SNOM 360 with extension panel is one of the best options, it handles all the extension indication status and has enough line extensions to cover up to 54 extensions. Only the Polycom 601 comes close. Regards Rob On 2/8/06, Garth van Sittert < garth@bitco.co.za <mailto:garth@bitco.co...

Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics & Computer Science) ----------------- Mobile: +27 (0)83 791 6662 Email: garth@bitco.co.za Phone: 08600 BITCO Web: www.bitco.co.za

...el like %${number}) exten => s,6,MySQL(Connect connid hostname username password dbname) exten => s,7,MySQL(Query resultid ${connid} ${query}) But there seems to be a problem with the % sign and I don't know how to hash it out. It works without the % sign. Thanks Kind Regards Garth

Hi All Has anyone come across a handset that can somehow replace FOP? Some users don't like FOP unless it is on a dedicated PC. Thanks Garth

...(auth any, allow *), the agent can successfully retrieve his catalog and plugins and apply them. Using webrick works fine as well. The agent shows this: err: /File[/var/lib/puppet/lib]: Failed to generate additional resources using ''eval_generate: Error 403 on SERVER: Forbidden request: garth.cc.gatech.edu(130.207.127.12) access to /file_metadata/plugins [search] at line 0 err: /File[/var/lib/puppet/lib]: Could not evaluate: Error 403 on SERVER: Forbidden request: garth.cc.gatech.edu(130.207.127.12) access to /file_metadata/plugins [find] at line 0 Could not retrieve file metadata for...

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.

...terisk and Asterisk and the Legacy PBX. At what level does bridging happen? Do the streams get passed directly between to two PRI ports? How would recording effect this? Tap into the streams etc? If someone could explain a bit more on the architecture around this, that would be great. Thanks Garth

Hi All Does anyone know how the hints in asterisk works? How does a SIP phone interact with the hints? I am having a problem with certain phone models that do not set the hints correctly when I list the hints with a 'show hints'. Thanks Garth

...he win2k machines claim to be "browser servers", unless I set the "maintain_browse_list" registry key to be false, but AFAIK this is just setting if the machine can become a local browse master and is not related to the bizzare behaviour we are seeing above. Any ideas? Thanks, Garth. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 232 bytes Desc: not available Url : http://lists.samba.org/archive/samba/attachments/20011203/b6179517/attachment.bin

...the CallerID? Have you checked with your provider that they've enabled callerid? If yes, are you using a correct number that the provider allows? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Garth van Sittert Skickat: den 2 februari 2006 12:37 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: [Asterisk-Users] Outbound Caller ID number on E1 Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearin...

...hen I tried moving it to Centos 5 I experienced a complete OS crash when calling over HFC misdn channels. Didn't really have time to investigate and dumps as it was a live machine. I have tried updating all relevant asterisk software, but to no avail. Anyone have any ideas? Kind Regards Garth

Anybody tried one with Asterisk yet ? Views ? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration

...the Samba DC when I attempt to run dcpromo and it throws a nasty warning about AD data being lost. If I run Sites and Services, both DCs show up and are viewable. Also, in Users and Computers, both DCs correctly show up in Domain Controllers. Not sure what to do next. Help appreciated. Thanx, Garth

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

...iving into discussion what is good for security and how it affects that. That would be what I would recommend to myself (which I did way back...). If I were choosing the book to get good start today, I would choose: UNIX and Linux System Administration Handbook (4th Edition) 2010 by Evi Nemeth and Garth Snyder - don't worry about "outdated...", remember, you first need fundamentals. It is not as "fundamental" as some of the books of the past I remember, but I'd rather mention it that those really old books. I'm sure, someone may suggest better book, maybe even fre...

...ot should create the needed subdir since I can?t find anything in the docs to suggest a way to create it otherwise. I?ve tried all I can think of at the moment to try to remedy this without success. I?m happy to provide additional details as needed to try to track this down. Thanks in advance, Garth -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3730 bytes Desc: not available URL: <https://dovecot.org/pipermail/dovecot/attachments/20171217/69144404/attachment-0001.p7s>

...James HR 54223 3 Howe Finance 80000 4 Jones Finance 82000 5 Norwood IT 67000 6 Benson Sales 76000 7 Smith Sales 65778 8 Baker HR 56778 9 Dempsey HR 78999 10 Nolan Sales 45667 11 Garth Finance 89777 12 Jameson IT 56786 I want to calculate both the mean salary broken down by Department and also the total amount paid out per department i.e. I want both sum(Salary) and mean(Salary) for each Department. Right now, I am using aggregate.data.frame twice, creatin...

...not much else. I have occasionally checked to see if it has become a simple process of moving FSMO roles and demoting the 2003 AD server but it's not clear from the forum or the docs. Has it become that simple or does it still require a fair bit of fiddling around to make it work? Thanx, Garth

...nfiguration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth

...ut it was not. I would love to know when this bug fix will be incorporated into a release, and it would be nice to know what needed to change. If it's a simple fix I would just make the change and recompile the voicemail app rather than wait for a release. Can anyone give me some more info? Garth -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060508/50c88c21/attachment.htm