[Asterisk-Users] Re: Automathic call forwarding (Gianni (priv.))
greennet.ge
oleg at greennet.ge
Sun Oct 30 22:38:20 MST 2005
Здравствуйте, asterisk-users-request.
Вы писали 30 октября 2005 г., 21:00:14:
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> Today's Topics:
> 1. Automathic call forwarding (Gianni (priv.))
> 2. Re: SPA3000 as trunk - no caller ID (Ben Higley)
> 3. Re: feature usage/digit detection (Bill Michaelson)
> 4. Re: Re: feature usage/digit detection (Andrew Kohlsmith)
> 5. RE: SPA3000 as trunk - no caller ID (Anders Svensson)
> 6. Re: libpri (Mark Quitoriano)
> 7. Re: libpri (Michael Bielicki)
> 8. Re: VoiceMailMain() in 1.2-beta (Leif Madsen)
> 9. RE: SPA3000 as trunk - no caller ID (Ben Higley)
> 10. RE: SCCP support is making good progress (Chris Bagnall)
> 11. RE: Webui to show registered phones (Paul)
> 12. RE: SPA3000 as trunk - no caller ID (Anders Svensson)
> 13. Re: SCCP support is making good progress (Zoa)
> 14. no sip peers after restarting asterisk? (Rich Adamson)
> 15. Re: Re: feature usage/digit detection (Eric "ManxPower" Wieling)
> 16. Re: SCCP support is making good progress (Stefan Gofferje)
> 17. Re: no sip peers after restarting asterisk? (Kevin P. Fleming)
> 18. Re: no sip peers after restarting asterisk? (Andrew Kohlsmith)
> 19. Re: no sip peers after restarting asterisk? (Rich Adamson)
> ----------------------------------------------------------------------
> Message: 1
> Date: Sun, 30 Oct 2005 16:27:21 +0100
> From: "Gianni \(priv.\)" <gianni at gminetti.net>
> Subject: [Asterisk-Users] Automathic call forwarding
> To: <asterisk-users at lists.digium.com>
> Message-ID: <000001c5dd66$6c63f520$07a1a8c0 at Saturno>
> Content-Type: text/plain; charset="us-ascii"
> Hello.
> I wonder if someone cal help me to find the right way to implement the below
> described TO-BE scenario (basically automatic farwarding from incoming
> calls).
> *** Background:
> - a VoIP/PSTN gateway Mediatrix 1104 registers on Asterisk at Home as UAs from
> 301 to 304. This Mediatrix is the gateway (4 port FXS) between a SIP/VoIP
> domain and a legacy PBX Nortel Meridian 1.
> - others UA (SIP/VoIP terminals extension from 100 to 140) also register
> into Asterisk at home
> *** AS-IS situation
> 1) UA 100 dial let's say 301 and get a PSTN line from the Mediatrix
> (mediatrix is then connected by FSX/FXO to a Nortel Meridian 1)
> 2) If another UA, let's say 101 wants to have a PSTN line, it should now
> that 301 is busy because of 100 in progress call and therefore it shall
> call. let's say 302 (likely after having found 301 busy)
> 3) And so on...
> *** TO-BE scenario (to be achieved)
> 1) UA 301 to 304 (Mediatrix VoIP gateway registered UA) are logically
> grouped and referred by a virtual extesion, let's say 999
> 2) any UA from VoIP domain calls 999 and Asterisk automatically route the
> incoming call on the first available line or if not, put it on hold.
> Something like
> IF port 301 is busy THEN reroute call on 302
> IF port 302 is busy THEN reroute call on 303
> IF port 303 is busy THEN reroute call on 303
> IF port 304 is busy THEN put on hold for x minutes
> Thanks in advance for your help
> Gianni
> ------------------------------
> Message: 2
> Date: Sun, 30 Oct 2005 07:12:06 -0800 (PST)
> From: "Ben Higley" <pbx at itsngroup.com>
> Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID
> To: john at argv.co.uk, "Asterisk Users Mailing List - Non-Commercial
> Discussion" <asterisk-users at lists.digium.com>
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <1235.192.168.1.141.1130685126.squirrel at mail.itsngroup.com>
> Content-Type: text/plain;charset=iso-8859-1
> I am trying to do the SIPURA-3000 as an incoming pstn trunk. I found the
> how-to on the geekgazette as well, however, my sipura-3000 only just sits
> and rings and rings and rings. I have set up the peer and the user values,
> as per the configuration, and when I look at the web status info page of
> the spa3000 it just says ringing ringing ringing. If I turn on
> ring-thru-to-line-1 and set it to yes, then the phone rings, but i cannot
> for the life of me get it to go into the extension that i have defined on
> the asterisk system.
> Could someone assist me with this?
> Thanks.
>> Kerry Garrison wrote:
>>> A phone plugged into it will grab the CID on about the second ring and I
>>> have adjusted the SPA3000 out to 5 rings with no difference. What gets
>>> passed to asterisk is whatever is set in the 3000's Display Name field.
>>> If
>>> the Display Name field is blank, then nothing comes across and the
>>> phones
>>> display 'Unknown'. I have been wondering if there is a variable you can
>>> put
>>> into the display field. There are some fields that use variables like
>>> $PROXY
>>> and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID.
>>
>> You don't need any clever manipulation tricks with the current firmware.
>> Have you got PSTN CID for VOIP CID set to yes ?
>>
>> jd
>>
>> --
>>
>> John Daragon john at argv.co.uk
>> argv[0] limited
>> Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
>> v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
> ------------------------------
> Message: 3
> Date: Sun, 30 Oct 2005 11:13:32 -0500
> From: Bill Michaelson <bill at cosi.com>
> Subject: [Asterisk-Users] Re: feature usage/digit detection
> To: asterisk-users at lists.digium.com
> Message-ID: <4364F12C.5020200 at cosi.com>
> Content-Type: text/plain; charset="us-ascii"
> Thanks for the answer. Doesn't solve my problem, but that's only
> because I didn't state my goal. You have corrected a misconseption
> on my part, which ought to get me closer. I'll explain...
> Indeed, I do have the "tT" options in the dial command. This is
> because I thought this would enable the use of the '#' for
> transfers, and it works satisfactorily. I also have various '*N'
> definitions in features.conf, but these don't work. I suppose I do
> have to rethink my strategy as you've suggested, but I don't know
> how to have my cake and eat it.. (?)
> By the way, I am using various SIP phones, with various DTMF
> detection techniques (e.g. ZyXEL wifi:inband, Grandstream BT101 and
> ATA-488:INFO) with apparent success because many features do work
> (such as transfer with #).
> Message: 22
> Date: Sun, 30 Oct 2005 10:57:57 -0400
> From: Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com>
> Subject: Re: [Asterisk-Users] gotta be a dumb question...
> To: asterisk-users at lists.digium.com
> Message-ID: <200510300957.57769.akohlsmith-asterisk at benshaw.com>
> Content-Type: text/plain; charset="iso-8859-1"
> On Sunday 30 October 2005 09:44, Bill Michaelson wrote:
>>> -- Attempting native bridge of SIP/215-b09e and SIP/259412-5967
>>>
>>> Now, I've got canreinvite=no in every sip definition, but it happens
>>> anyway.
>>
>>
> That has nothing to do with reinvites.
> In Asterisk terms, a native bridge between two channels is the lowest-latency
> connection between those channels without dropping out of the loop entirely.
> Essentially a native bridge just reads voice frames from one and transmits
> them to the other. There is no codec translation or any other goodness going
> on.
> When you hit a DTMF digit (you must be using inband DTMF here I think), the
> native bridge must be dropped because Asterisk needs to prepare to do
> something with the DTMF (transfer, etc.) -- when Asterisk has determined that
> it doesn't need to do anything special, it sets up the native bridge again to
> minimize the latency once again.
> The fact that your * is getting "swallowed" tells me that you are using * in
> features.conf to denote special keypresses to Asterisk. In Dial() you likely
> have the 't' or 'T' flags set, which causes Asterisk to "think" that those
> DTMF digits are for it, not for the other side. Either edit features.conf,
> remove the 't' or 'T' flags from the Dial() command or rethink your strategy.
> I hope this is an acceptable answer, and I certainly hope it's accurate. It's
> my understanding of the system anyway. If you prefer not to have these
> types of messages, you need to turn DOWN the verbosity level.
> -A.
> -------------- next part --------------
> Skipped content of type multipart/related
> ------------------------------
> Message: 4
> Date: Sun, 30 Oct 2005 12:36:28 -0400
> From: Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com>
> Subject: Re: [Asterisk-Users] Re: feature usage/digit detection
> To: asterisk-users at lists.digium.com
> Message-ID: <200510301136.28176.akohlsmith-asterisk at benshaw.com>
> Content-Type: text/plain; charset="iso-8859-1"
> On Sunday 30 October 2005 11:13, Bill Michaelson wrote:
>> Indeed, I do have the "tT" options in the dial command. This is because I
>> thought this would enable the use of the '#' for transfers, and it works
>> satisfactorily. I also have various '*N' definitions in features.conf, but
>> these don't work. I suppose I do have to rethink my strategy as you've
>> suggested, but I don't know how to have my cake and eat it.. (?)
> That's exactly what the 't' and 'T' options do, just make sure you are using
> the right one, I find it almost NEVER desireable to have both. 'T' allows
> the calling user to transfer with '#', 't' allows the called user to do so.
> if you're dialing between extensions in an office, you want both, but most
> other times you want one or the other.
> If I'm not mistaken only 'pbx' threads can make use of the other features in
> features.conf. tT is only for features in the [featuremap] section of
> features.conf. I think. (blind/attended transfers, call record, disconnect,
> etc.)
> I think. :-)
> -A.
> ------------------------------
> Message: 5
> Date: Sun, 30 Oct 2005 17:43:27 +0100
> From: "Anders Svensson" <anders at bobascom.com>
> Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20051030164323.C064037E42 at smtp4-2-sn2.hy.skanova.net>
> Content-Type: text/plain; charset="us-ascii"
> Have you read this?
> http://voipspeak.net/index.php?option=c . d=99999999
> Anders
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ben Higley
> Sent: den 30 oktober 2005 16:12
> To: john at argv.co.uk; Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID
> I am trying to do the SIPURA-3000 as an incoming pstn trunk. I found the
> how-to on the geekgazette as well, however, my sipura-3000 only just sits
> and rings and rings and rings. I have set up the peer and the user values,
> as per the configuration, and when I look at the web status info page of
> the spa3000 it just says ringing ringing ringing. If I turn on
> ring-thru-to-line-1 and set it to yes, then the phone rings, but i cannot
> for the life of me get it to go into the extension that i have defined on
> the asterisk system.
> Could someone assist me with this?
> Thanks.
>> Kerry Garrison wrote:
>>> A phone plugged into it will grab the CID on about the second ring and I
>>> have adjusted the SPA3000 out to 5 rings with no difference. What gets
>>> passed to asterisk is whatever is set in the 3000's Display Name field.
>>> If
>>> the Display Name field is blank, then nothing comes across and the
>>> phones
>>> display 'Unknown'. I have been wondering if there is a variable you can
>>> put
>>> into the display field. There are some fields that use variables like
>>> $PROXY
>>> and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID.
>>
>> You don't need any clever manipulation tricks with the current firmware.
>> Have you got PSTN CID for VOIP CID set to yes ?
>>
>> jd
>>
>> --
>>
>> John Daragon john at argv.co.uk
>> argv[0] limited
>> Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
>> v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
> Asterisk-Users mailing list
> Asterisk-Users at 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
> ------------------------------
> Message: 6
> Date: Mon, 31 Oct 2005 00:51:54 +0800
> From: Mark Quitoriano <markquitoriano at gmail.com>
> Subject: Re: [Asterisk-Users] libpri
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
>
> <6b542ec90510300851m6d2ccdf2y14e5052186ca626b at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
> ok tnx guys.
> On 10/30/05, Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com> wrote:
>>
>> On Sunday 30 October 2005 09:48, Michael Bielicki wrote:
>> > no, libpri is only needed for pri trunks
>>
>> It's also needed for ISDN BRI, I think...
>>
>> Certainly not for analog FXS or FXO though, you're right.
>>
>> -A.
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com <http://Easynews.com>--
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
> --
> Regards,
> Mark Quitoriano, CCNA
> http://www.atamanetworks.com
> Fan the flame...
> http://www.spreadfirefox.com/?q=user/register&r=19441
> -------------- next part --------------
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> ------------------------------
> Message: 7
> Date: Sun, 30 Oct 2005 17:55:52 +0100
> From: Michael Bielicki <cypromis at gmail.com>
> Subject: Re: [Asterisk-Users] libpri
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
>
> <18fec2710510300855w72147303r443d3e0b65929a07 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
> for BRI only if you have patched it for bristuff :)
> On 10/30/05, Mark Quitoriano <markquitoriano at gmail.com> wrote:
>>
>> ok tnx guys.
>>
>> On 10/30/05, Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com> wrote:
>> >
>> > On Sunday 30 October 2005 09:48, Michael Bielicki wrote:
>> > > no, libpri is only needed for pri trunks
>> >
>> > It's also needed for ISDN BRI, I think...
>> >
>> > Certainly not for analog FXS or FXO though, you're right.
>> >
>> > -A.
>> > _______________________________________________
>> > --Bandwidth and Colocation sponsored by
>> Easynews.com<http://Easynews.com>--
>> >
>> > Asterisk-Users mailing list
>> > Asterisk-Users at 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
>> >
>>
>>
>>
>> --
>> Regards,
>> Mark Quitoriano, CCNA
>> http://www.atamanetworks.com
>>
>> Fan the flame...
>> http://www.spreadfirefox.com/?q=user/register&r=19441
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com <http://Easynews.com>--
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
>>
> --
> Michal Bielicki
> Halo Kwadrat Sp. z o.o.
> http://www.asterisk.pl/
> http://www.openpbx.org/
> -------------- next part --------------
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> ------------------------------
> Message: 8
> Date: Sun, 30 Oct 2005 11:57:37 -0500
> From: Leif Madsen <asterisk.leif.madsen at gmail.com>
> Subject: Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <c485d190510300857i6f687a1ej431ea916a3383db6 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
> On 10/30/05, David Bandel <david.bandel at gmail.com> wrote:
>> Have the OReilley book. Also the new 1.2 book from asteriskdocs.org.
> Psssst... they're the same book :)
> --
> Leif Madsen - http://www.leifmadsen.com
> http://www.asteriskdocs.org -- Co-Founder
> http://www.oreilly.com/catalog/asterisk -- Co-Author
> ------------------------------
> Message: 9
> Date: Sun, 30 Oct 2005 09:07:06 -0800 (PST)
> From: "Ben Higley" <pbx at itsngroup.com>
> Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <1686.192.168.1.141.1130692026.squirrel at mail.itsngroup.com>
> Content-Type: text/plain;charset=iso-8859-1
> That link is not found....
>> Have you read this?
>>
>> http://voipspeak.net/index.php?option=c . d=99999999
>>
>> Anders
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ben Higley
>> Sent: den 30 oktober 2005 16:12
>> To: john at argv.co.uk; Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID
>>
>>
>> I am trying to do the SIPURA-3000 as an incoming pstn trunk. I found the
>> how-to on the geekgazette as well, however, my sipura-3000 only just sits
>> and rings and rings and rings. I have set up the peer and the user values,
>> as per the configuration, and when I look at the web status info page of
>> the spa3000 it just says ringing ringing ringing. If I turn on
>> ring-thru-to-line-1 and set it to yes, then the phone rings, but i cannot
>> for the life of me get it to go into the extension that i have defined on
>> the asterisk system.
>>
>> Could someone assist me with this?
>>
>> Thanks.
>>
>>> Kerry Garrison wrote:
>>>> A phone plugged into it will grab the CID on about the second ring and
>>>> I
>>>> have adjusted the SPA3000 out to 5 rings with no difference. What gets
>>>> passed to asterisk is whatever is set in the 3000's Display Name field.
>>>> If
>>>> the Display Name field is blank, then nothing comes across and the
>>>> phones
>>>> display 'Unknown'. I have been wondering if there is a variable you can
>>>> put
>>>> into the display field. There are some fields that use variables like
>>>> $PROXY
>>>> and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID.
>>>
>>> You don't need any clever manipulation tricks with the current firmware.
>>> Have you got PSTN CID for VOIP CID set to yes ?
>>>
>>> jd
>>>
>>> --
>>>
>>> John Daragon john at argv.co.uk
>>> argv[0] limited
>>> Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
>>> v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
>>>
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation sponsored by Easynews.com --
>>>
>>> Asterisk-Users mailing list
>>> Asterisk-Users at 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
>>>
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
> ------------------------------
> Message: 10
> Date: Sun, 30 Oct 2005 17:04:22 -0000
> From: "Chris Bagnall" <asterisk at minotaur.cc>
> Subject: RE: [Asterisk-Users] SCCP support is making good progress
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <E1EWGbd-000668-1t at tethys.minotaur.uk.net>
> Content-Type: text/plain; charset="US-ASCII"
>> whoever owns a Cisco phone and is unhappy about slow
>> firmware, incomplete XML support etc... should really have a
>> look at Sergio Chersovani's rewrite of chan-sccp!
> Is there a good resource out there for people who don't have a lot of
> experience with Cisco phones? I picked up a 7960 earlier this week to give
> potential clients an example of what they get when they spend a *lot* of
> money on IP phones, but I must confess I'm having a nightmare of a time
> trying to configure it.
> The main problem seem to be that I have nothing but a phone and a brief
> licence agreement/regulatory approval sheet, and nothing else. I've trawled
> through the numerous pages about these phones both on Cisco's website and on
> voip-info, but I'm still not really sure what files I need to have on the
> TFTP server to get the phone going in the first place, or find some
> up-to-date examples to work from. Even after that I'm not sure I'll be able
> to upgrade the firmware without a Cisco service agreement (from what I've
> read), which is ridiculous for a phone that's twice as expensive as many
> other enterprise IP phones.
> Any suggested reading others on the list have found helpful in this
> scenario?
> Thanks in advance.
> Regards,
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> This email is made from 100% recycled electrons
> ------------------------------
> Message: 11
> Date: Sun, 30 Oct 2005 12:06:31 -0500
> From: Paul <paul at siliconvp.com>
> Subject: RE: [Asterisk-Users] Webui to show registered phones
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> <asterisk-users at lists.digium.com>
> Message-ID: <0IP600FUUNIL9VZP at mta10.srv.hcvlny.cv.net>
> Content-Type: text/plain; charset=iso-8859-1
> Is this release under the GPL?
> I see no mention of this windows based program on your web site.
> ::)
> Paul
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
>> bounces at lists.digium.com] On Behalf Of Saul Diaz
>> Sent: Saturday, October 29, 2005 11:08 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] Webui to show registered phones
>>
>> Hi
>>
>> For those who are insterested in monitoring and managing easilly the
>> asterisk server..
>>
>> this is a solution for multitenant hosted PBX o single tenant is windows
>> based (the admin of couse) and
>>
>> http://www.cripiland.com/screenshots/manager3.jpg
>> http://www.cripiland.com/screenshots/manager4.jpg
>> http://www.cripiland.com/screenshots/manager1.jpg
>> http://www.cripiland.com/screenshots/manager2.jpg
>>
>> regards
>> Saul
>>
>> Matt Gibson wrote:
>>
>> > Hi Guys,
>> >
>> > Here's what I use to view the current IAX and SIP peer status. It's
>> > not very pretty, but it works.
>> > I also have an included script (vm.php) that will show the current
>> > voicemail usage for a box.
>> >
>> > Uses php asterisk library to work through asterisk manager.
>> >
>> > Configure your options in cfg.php
>> >
>> > Matt
>> >
>> >
>> > Nicolбs Gudiсo wrote:
>> >
>> >>> Hi all, does anyone know if there is any app/webui that can show
>> phones
>> >>> that are currently registered to *. I guess this sort of funcionality
>> >>> counld be grabbed from the CLI with iax2 show peers and sip show
>> peers,
>> >>> but having little programming knowledge wouldn't know where to start.
>> >>>
>> >>> I'm asking because we currently have several sip phones onsite and
>> lots
>> >>> of remote iax2 users who would like to see availability without
>> >>> dialing.
>> >>>
>> >>
>> >>
>> >> <plug>You can try with the Flash Operator Panel</plug>
>> >> http://www.asternic.org , it does all sort of things including sip and
>> >> iax availability (you have to enable qualify for them). Regards,
>> >>
>> >> --
>> >> Nicolбs Gudiсo
>> >> Buenos Aires - Argentina
>> >> _______________________________________________
>> >> --Bandwidth and Colocation sponsored by Easynews.com --
>> >>
>> >> Asterisk-Users mailing list
>> >> Asterisk-Users at 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
>> >>
>> >
>> >------------------------------------------------------------------------
>> >
>> >_______________________________________________
>> >--Bandwidth and Colocation sponsored by Easynews.com --
>> >
>> >Asterisk-Users mailing list
>> >Asterisk-Users at 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
>> >
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
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> ------------------------------
> Message: 12
> Date: Sun, 30 Oct 2005 18:09:06 +0100
> From: "Anders Svensson" <anders at bobascom.com>
> Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20051030170903.46D2037E42 at smtp4-2-sn2.hy.skanova.net>
> Content-Type: text/plain; charset="us-ascii"
> http://voipspeak.net/index.php?option=com_content&task=view&id=24&Itemid=27
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ben Higley
> Sent: den 30 oktober 2005 18:07
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID
> That link is not found....
>> Have you read this?
>>
>> http://voipspeak.net/index.php?option=c . d=99999999
>>
>> Anders
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ben Higley
>> Sent: den 30 oktober 2005 16:12
>> To: john at argv.co.uk; Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID
>>
>>
>> I am trying to do the SIPURA-3000 as an incoming pstn trunk. I found the
>> how-to on the geekgazette as well, however, my sipura-3000 only just sits
>> and rings and rings and rings. I have set up the peer and the user values,
>> as per the configuration, and when I look at the web status info page of
>> the spa3000 it just says ringing ringing ringing. If I turn on
>> ring-thru-to-line-1 and set it to yes, then the phone rings, but i cannot
>> for the life of me get it to go into the extension that i have defined on
>> the asterisk system.
>>
>> Could someone assist me with this?
>>
>> Thanks.
>>
>>> Kerry Garrison wrote:
>>>> A phone plugged into it will grab the CID on about the second ring and
>>>> I
>>>> have adjusted the SPA3000 out to 5 rings with no difference. What gets
>>>> passed to asterisk is whatever is set in the 3000's Display Name field.
>>>> If
>>>> the Display Name field is blank, then nothing comes across and the
>>>> phones
>>>> display 'Unknown'. I have been wondering if there is a variable you can
>>>> put
>>>> into the display field. There are some fields that use variables like
>>>> $PROXY
>>>> and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID.
>>>
>>> You don't need any clever manipulation tricks with the current firmware.
>>> Have you got PSTN CID for VOIP CID set to yes ?
>>>
>>> jd
>>>
>>> --
>>>
>>> John Daragon john at argv.co.uk
>>> argv[0] limited
>>> Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
>>> v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
>>>
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation sponsored by Easynews.com --
>>>
>>> Asterisk-Users mailing list
>>> Asterisk-Users at 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
>>>
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
> Asterisk-Users mailing list
> Asterisk-Users at 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
> ------------------------------
> Message: 13
> Date: Sun, 30 Oct 2005 19:10:22 +0200
> From: Zoa <zoachien at securax.org>
> Subject: Re: [Asterisk-Users] SCCP support is making good progress
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4364FE7E.6010601 at securax.org>
> Content-Type: text/plain; charset="iso-8859-1"
> Have a look here, :
> http://www.asteriskguru.com/tutorials/cisco_7960_skinny_chan_sccp.html
> If you find any other suggestions, remarks after installing, please post
> them as a comment to the page.
> Zoa
> Chris Bagnall wrote:
>>>whoever owns a Cisco phone and is unhappy about slow
>>>firmware, incomplete XML support etc... should really have a
>>>look at Sergio Chersovani's rewrite of chan-sccp!
>>>
>>>
>>
>>Is there a good resource out there for people who don't have a lot of
>>experience with Cisco phones? I picked up a 7960 earlier this week to give
>>potential clients an example of what they get when they spend a *lot* of
>>money on IP phones, but I must confess I'm having a nightmare of a time
>>trying to configure it.
>>
>>The main problem seem to be that I have nothing but a phone and a brief
>>licence agreement/regulatory approval sheet, and nothing else. I've trawled
>>through the numerous pages about these phones both on Cisco's website and on
>>voip-info, but I'm still not really sure what files I need to have on the
>>TFTP server to get the phone going in the first place, or find some
>>up-to-date examples to work from. Even after that I'm not sure I'll be able
>>to upgrade the firmware without a Cisco service agreement (from what I've
>>read), which is ridiculous for a phone that's twice as expensive as many
>>other enterprise IP phones.
>>
>>Any suggested reading others on the list have found helpful in this
>>scenario?
>>
>>Thanks in advance.
>>
>>Regards,
>>
>>Chris
>>
>>
> -------------- next part --------------
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> ------------------------------
> Message: 14
> Date: Sun, 30 Oct 2005 11:09:32 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: [Asterisk-Users] no sip peers after restarting asterisk?
> To: Asterisk-users-list <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1130692390.adar0 at vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
> Just updated cvs-head this morning, and now on a 'stop now' and restart,
> * doesn't know about the previously registered sip phones (as shown with
> sip show peers) on fc3.
> Once the phones register again, they can be called, but not until then.
> Not sure what's going on yet... anyone seeing the same?
> ------------------------------
> Message: 15
> Date: Sun, 30 Oct 2005 11:17:41 -0600
> From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
> Subject: Re: [Asterisk-Users] Re: feature usage/digit detection
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <43650035.5030304 at fnords.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> Andrew Kohlsmith wrote:
>> On Sunday 30 October 2005 11:13, Bill Michaelson wrote:
>>
>>>Indeed, I do have the "tT" options in the dial command. This is because I
>>>thought this would enable the use of the '#' for transfers, and it works
>>>satisfactorily. I also have various '*N' definitions in features.conf, but
>>>these don't work. I suppose I do have to rethink my strategy as you've
>>>suggested, but I don't know how to have my cake and eat it.. (?)
>>
>>
>> That's exactly what the 't' and 'T' options do, just make sure you are using
>> the right one, I find it almost NEVER desireable to have both. 'T' allows
>> the calling user to transfer with '#', 't' allows the called user to do so.
>> if you're dialing between extensions in an office, you want both, but most
>> other times you want one or the other.
>>
>> If I'm not mistaken only 'pbx' threads can make use of the other features in
>> features.conf. tT is only for features in the [featuremap] section of
>> features.conf. I think. (blind/attended transfers, call record, disconnect,
>> etc.)
> T/t with # are in 1.0.x and later. The other features, like changing
> the # to something else and the other features are only available in
> 1.2beta and CVS-HEAD.
> ------------------------------
> Message: 16
> Date: Sun, 30 Oct 2005 18:20:19 +0100
> From: Stefan Gofferje <stefan at gofferje.homelinux.org>
> Subject: Re: [Asterisk-Users] SCCP support is making good progress
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <436500D3.7040909 at gofferje.homelinux.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> Chris Bagnall schrieb:
>>>whoever owns a Cisco phone and is unhappy about slow
>>>firmware, incomplete XML support etc... should really have a
>>>look at Sergio Chersovani's rewrite of chan-sccp!
>>
>>
>> Is there a good resource out there for people who don't have a lot of
>> experience with Cisco phones? I picked up a 7960 earlier this week to give
>> potential clients an example of what they get when they spend a *lot* of
>> money on IP phones, but I must confess I'm having a nightmare of a time
>> trying to configure it.
>>
>> The main problem seem to be that I have nothing but a phone and a brief
>> licence agreement/regulatory approval sheet, and nothing else. I've trawled
>> through the numerous pages about these phones both on Cisco's website and on
>> voip-info, but I'm still not really sure what files I need to have on the
>> TFTP server to get the phone going in the first place, or find some
>> up-to-date examples to work from. Even after that I'm not sure I'll be able
>> to upgrade the firmware without a Cisco service agreement (from what I've
>> read), which is ridiculous for a phone that's twice as expensive as many
>> other enterprise IP phones.
>>
>> Any suggested reading others on the list have found helpful in this
>> scenario?
> The list archives of chan-sccp-users provides a lot of information.
> www.voip-info.org also has. There are a number of ressources at
> cisco.com and if all this does not help, the people at chan-sccp-users
> or forum.chan-sccp.org use to friendly answer questions.
> There are also a number of people working at various howtos at the moment.
> Regards,
> Stefan
U have to use Queue read queue.conf & agents.conf
Just make your extensions 301-304 agents & 999 will be queue number.
--
С уважением,
greennet.ge mailto:oleg at greennet.ge
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