greennet.ge
2005-Oct-30 22:38 UTC
[Asterisk-Users] Re: Automathic call forwarding (Gianni (priv.))
????????????, asterisk-users-request. ?? ?????? 30 ??????? 2005 ?., 21:00:14:> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com> To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com> You can reach the person managing the list at > asterisk-users-owner@lists.digium.com> When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..."> 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@gminetti.net> > Subject: [Asterisk-Users] Automathic call forwarding > To: <asterisk-users@lists.digium.com> > Message-ID: <000001c5dd66$6c63f520$07a1a8c0@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@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@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@itsngroup.com> > Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID > To: john@argv.co.uk, "Asterisk Users Mailing List - Non-Commercial > Discussion" <asterisk-users@lists.digium.com> > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > <1235.192.168.1.141.1130685126.squirrel@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@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@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@cosi.com> > Subject: [Asterisk-Users] Re: feature usage/digit detection > To: asterisk-users@lists.digium.com > Message-ID: <4364F12C.5020200@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@benshaw.com> > Subject: Re: [Asterisk-Users] gotta be a dumb question... > To: asterisk-users@lists.digium.com > Message-ID: <200510300957.57769.akohlsmith-asterisk@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@benshaw.com> > Subject: Re: [Asterisk-Users] Re: feature usage/digit detection > To: asterisk-users@lists.digium.com > Message-ID: <200510301136.28176.akohlsmith-asterisk@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@bobascom.com> > Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <20051030164323.C064037E42@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@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ben Higley > Sent: den 30 oktober 2005 16:12 > To: john@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@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@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@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@gmail.com> > Subject: Re: [Asterisk-Users] libpri > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > > <6b542ec90510300851m6d2ccdf2y14e5052186ca626b@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1"> ok tnx guys.> On 10/30/05, Andrew Kohlsmith <akohlsmith-asterisk@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@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 -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20051031/2a7ab05b/attachment-0001.htm> ------------------------------> Message: 7 > Date: Sun, 30 Oct 2005 17:55:52 +0100 > From: Michael Bielicki <cypromis@gmail.com> > Subject: Re: [Asterisk-Users] libpri > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > > <18fec2710510300855w72147303r443d3e0b65929a07@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@gmail.com> wrote: >> >> ok tnx guys. >> >> On 10/30/05, Andrew Kohlsmith <akohlsmith-asterisk@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@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@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 -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20051030/45c8e619/attachment-0001.htm> ------------------------------> Message: 8 > Date: Sun, 30 Oct 2005 11:57:37 -0500 > From: Leif Madsen <asterisk.leif.madsen@gmail.com> > Subject: Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > <c485d190510300857i6f687a1ej431ea916a3383db6@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1> On 10/30/05, David Bandel <david.bandel@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@itsngroup.com> > Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <1686.192.168.1.141.1130692026.squirrel@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@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ben Higley >> Sent: den 30 oktober 2005 16:12 >> To: john@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@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@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@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@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@minotaur.cc> > Subject: RE: [Asterisk-Users] SCCP support is making good progress > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <E1EWGbd-000668-1t@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@siliconvp.com> > Subject: RE: [Asterisk-Users] Webui to show registered phones > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > <asterisk-users@lists.digium.com> > Message-ID: <0IP600FUUNIL9VZP@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@lists.digium.com [mailto:asterisk-users- >> bounces@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 >> > >> > >> > Nicol?s Gudi?o 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, >> >> >> >> -- >> >> Nicol?s Gudi?o >> >> Buenos Aires - Argentina >> >> _______________________________________________ >> >> --Bandwidth and Colocation sponsored by Easynews.com -- >> >> >> >> 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 >> >> >> > >> >------------------------------------------------------------------------ >> > >> >_______________________________________________ >> >--Bandwidth and Colocation sponsored by Easynews.com -- >> > >> >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 >> > >> >> _______________________________________________ >> --Bandwidth and Colocation sponsored by Easynews.com -- >> >> 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> ------------------------------> Message: 12 > Date: Sun, 30 Oct 2005 18:09:06 +0100 > From: "Anders Svensson" <anders@bobascom.com> > Subject: RE: [Asterisk-Users] SPA3000 as trunk - no caller ID > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <20051030170903.46D2037E42@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@lists.digium.com > [mailto:asterisk-users-bounces@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@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ben Higley >> Sent: den 30 oktober 2005 16:12 >> To: john@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@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@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@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@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@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@securax.org> > Subject: Re: [Asterisk-Users] SCCP support is making good progress > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4364FE7E.6010601@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 -------------- > A non-text attachment was scrubbed... > Name: signature.asc > Type: application/pgp-signature > Size: 254 bytes > Desc: OpenPGP digital signature > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20051030/73fcff2d/signature-0001.pgp> ------------------------------> Message: 14 > Date: Sun, 30 Oct 2005 11:09:32 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: [Asterisk-Users] no sip peers after restarting asterisk? > To: Asterisk-users-list <asterisk-users@lists.digium.com> > Message-ID: <Chameleon.1130692390.adar0@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@fnords.org> > Subject: Re: [Asterisk-Users] Re: feature usage/digit detection > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <43650035.5030304@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@gofferje.homelinux.org> > Subject: Re: [Asterisk-Users] SCCP support is making good progress > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <436500D3.7040909@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, > StefanU have to use Queue read queue.conf & agents.conf Just make your extensions 301-304 agents & 999 will be queue number. -- ? ?????????, greennet.ge mailto:oleg@greennet.ge