RE: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
The iMate is availble here, as well as the O2 XDAII. I have the O2 running here without too many issues..(appart from WiFi Sucking My Battery's will to live) With SJPhone seems to be mostly stable. The ECS-IAX PDA Client is WAY too unstable at the moment...conectivity/voice quality issues... Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 27 March 2006 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) I think the main issue for James and myself is that we can't buy anything in Australia. Paul Hales Technical Manager AsteriskIT - Original Message - From: "AR Tarzi" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, March 27, 2006 10:21 AM Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) > Not GSM/DECT but GSM/Wifi phones are available - This is not a > recommendation, I don't like what I've seen. > try www.imate.com (to start with) .. they have at least three types of GSM > phones that do Wifi .. They run windows so there are several sip softwares > and one IAX software that work with these - > > Also Nokia has a GSM phone that does Wifi but that's a symbian (OS) phone > (don't know of sip software that works with it). > > > - Original Message - > From: <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, March 27, 2006 00:48 > Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) > > > > If you find anything out, I would like to know. > > > > I have tried to find a gsm/wifi phone in the past (in melbourne) and > > failed. > > > > later, > > > > Paul Hales > > Technical Manager > > AsteriskIT > > > > - Original Message - > > From: "James Harper" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Saturday, March 25, 2006 11:21 AM > > Subject: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) > > > > > >> Now that I actually try and google for it, I can't find any dual mode > >> GSM/DECT handsets, only pages telling me that they exist without any > >> actual information!!! > >> > >> Does anyone know of any such handsets? (and even better, ones that are > >> available in Australia) I've searched a few of the major gsm > >> manufacturers (nokia, Panasonic, sonyericsson) but their web sites are > >> absolutely pathetic to the point being useless (or maybe I'm just in a > >> bad mood today :) > >> > >> Thanks > >> > >> James > >> > >> > -Original Message- > >> > From: [EMAIL PROTECTED] [mailto:asterisk-users- > >> > [EMAIL PROTECTED] On Behalf Of James Harper > >> > Sent: Friday, 24 March 2006 13:08 > >> > To: Asterisk Users Mailing List - Non-Commercial Discussion > >> > Subject: RE: [Asterisk-Users] Re: gsm picocells > >> > > >> > > Steve, > >> > > > >> > > Excellent explanation. > >> > > > >> > > In a nutshell, it might be better to just use a phone that can > >> > > automatically switch between GSM and WiFi. Of course, that's limited > >> > to > >> > > handful of handsets. > >> > > >> > I haven't done any sort of research, but I've been told that GSM+DECT > >> > phones are available, and while having them seamlessly switch network > >> > types during a call probably isn't possible, they can function as a > >> > cordless handset. > >> > > >> > Can anyone confirm or deny this? > >> > > >> > James > >> > ___ > >> > --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 > >> > >> ___ > >> --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 > >> > >> > > > > ___ > > --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 > > ___ > --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 > ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.3.2/293 - Release Date: 26/03/2006 -- No virus found in
RE: [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Thursday, 23 March 2006 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY) Erik Anderson wrote: > On 3/22/06, Andrew D Kirch <[EMAIL PROTECTED]> wrote: >> Andrew D Kirch >> Indianapolis, United States > > > Well if that isn't one of the most bizarre emails I've seen come > across this list. > > > -- > Erik Anderson > http://andersonfam.org Surprised SpamAssasin didn't pick it up on the way in ... :D -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.6/288 - Release Date: 22/03/2006 ___ --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] HFC USB (was MultiBRI in Australia - found one -maybe)
Hmm, > > I was using 0.3.0 rc24, or the unstable branch. I see 0.2.0 is listed as > 'stable' so maybe I should have used that. Please do keep me informed of > your progress. > > Craig After finally getting chan_misdn to load (missing #include to bitops.h under Debian at least) it still won't load, and won't tell me why even with all the debug stuff turned on. 0.3.0rc25 is what I'm using. chan_capi works in TE mode, but I can't get it working in NT mode which is what I want (keeps complaining about not being able to find a device for a blank msn). Could you please post something about what you did to get chan_misdn going? I have an idea that I've got a bad version of something compiled somewhere but hopefully it is solvable. James - OK Being the OH so Lazy person that I am...here are the steps that I took to get this all going. Started with my Stock Standard CentOS 4.2 install ... Installed 2.6.11 Kernel sources. Compiled and installed as per normal...turning off spinlock_debug and SMP Rebooted into new kernel. Installed mISDN using the install_misdn script Recompiled zaptel (for the hell of it...and so that I had a timming source) Manually setup the /etc/misdn-init.conf The autodiscovery thing didn't pickup the devices. Added the following three lines to my rc.local rmmod hfc_usb rmmod hisax /etc/init.d/misdn-init start Reboot once moreand that was it Dave -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.5/284 - Release Date: 17/03/2006 ___ --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] MultiBRI in Australia - found one - maybe
Faxing received by SpanDSP seems to work fine with these units. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Phelan Sent: Tuesday, 14 March 2006 9:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe HI Craig and all that is following this. I am running a Vanilla 2.6.11 >From cli, misdn show config Misdn General-Config: -> VERSION: 0.2.1 -> DEBUG_LEVEL: 1 -> TRACEFILE: not set -> TRACE_CALLS: false -> TRACE_DIR: /var/log/ -> BRIDGING: no-> STOP_TONE_AFTER_FIRST_DIGIT: yes -> APPEND_DIGITS2EXTEN: yes-> L1_INFO_OK: yes -> CLEAR_L3: no-> DYNAMIC_CRYPT: no -> CRYPT_PREFIX: **-> CRYPT_KEYS: test,muh So Far, no dropped calls etc Todays testing will be faxing. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Monday, 13 March 2006 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we haven't had any lockups but users are reporting dropped calls. Unfortunately for us this means dropping chan_mISDN in favour of the Cisco router containing BRI cards and then SIP from the Cisco to Asterisk. It may still be possible to use chan_capi with the mISDN drivers for the Drayteks but for us we've run out of time which is a bit of a bummer. I believe the problem is in chan_mISDN which is admittedly still an experimental driver at this stage with release candidates every few days for the past couple weeks. I'm still interested to know how you guys get along with these adapters. As I said, I think the problem is within chan_mISDN at this stage rather than in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware drivers or using chan_vISDN would be the way to go until chan_mISDN matures. Craig - Original Message - From: "James Harper" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, March 13, 2006 3:16 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe > > Got my 2 dreytek adapters today... > Dropped them on to my test system. After wadding thru my Memory of how to > setup mISDN, I had it up and running within about 2 hours. You might be receiving an email from me shortly then if I get stuck. If it wasn't for these annoying public holidays (Labour day in Victoria) mine would probably have arrived today too :) > Both of them operating in ptmp with no echo cancel turned on at this > stage. > Seems to be happy. That's quite comforting for initial testing. Could you try some faxing? And is there any way to measure latency with some hard figures, maybe by use of a repeater? Maybe something like this: Echo measurer -> BRI 1 -> BRI2 -> echo responder. Where the measurer dials the responder, sends out a ping, and measures the delay in the response. I find it hard to believe that any USB induced latency could be measurable in milliseconds... > Will drop them onto my local production box next week and see how we go :D Let us know! Thanks James ___ --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 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.1/279 - Release Date: 10/03/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.2/280 - Release Date: 13/03/2006 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.2/280 - Release Date: 13/03/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.2/280 - Release Date: 13/03/2006 ___ --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] MultiBRI in Australia - found one - maybe
HI Craig and all that is following this. I am running a Vanilla 2.6.11 >From cli, misdn show config Misdn General-Config: -> VERSION: 0.2.1 -> DEBUG_LEVEL: 1 -> TRACEFILE: not set -> TRACE_CALLS: false -> TRACE_DIR: /var/log/ -> BRIDGING: no-> STOP_TONE_AFTER_FIRST_DIGIT: yes -> APPEND_DIGITS2EXTEN: yes-> L1_INFO_OK: yes -> CLEAR_L3: no-> DYNAMIC_CRYPT: no -> CRYPT_PREFIX: **-> CRYPT_KEYS: test,muh So Far, no dropped calls etc Todays testing will be faxing. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Monday, 13 March 2006 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we haven't had any lockups but users are reporting dropped calls. Unfortunately for us this means dropping chan_mISDN in favour of the Cisco router containing BRI cards and then SIP from the Cisco to Asterisk. It may still be possible to use chan_capi with the mISDN drivers for the Drayteks but for us we've run out of time which is a bit of a bummer. I believe the problem is in chan_mISDN which is admittedly still an experimental driver at this stage with release candidates every few days for the past couple weeks. I'm still interested to know how you guys get along with these adapters. As I said, I think the problem is within chan_mISDN at this stage rather than in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware drivers or using chan_vISDN would be the way to go until chan_mISDN matures. Craig - Original Message - From: "James Harper" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, March 13, 2006 3:16 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe > > Got my 2 dreytek adapters today... > Dropped them on to my test system. After wadding thru my Memory of how to > setup mISDN, I had it up and running within about 2 hours. You might be receiving an email from me shortly then if I get stuck. If it wasn't for these annoying public holidays (Labour day in Victoria) mine would probably have arrived today too :) > Both of them operating in ptmp with no echo cancel turned on at this > stage. > Seems to be happy. That's quite comforting for initial testing. Could you try some faxing? And is there any way to measure latency with some hard figures, maybe by use of a repeater? Maybe something like this: Echo measurer -> BRI 1 -> BRI2 -> echo responder. Where the measurer dials the responder, sends out a ping, and measures the delay in the response. I find it hard to believe that any USB induced latency could be measurable in milliseconds... > Will drop them onto my local production box next week and see how we go :D Let us know! Thanks James ___ --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 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.1/279 - Release Date: 10/03/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.2/280 - Release Date: 13/03/2006 ___ --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] MultiBRI in Australia - found one - maybe
Got my 2 dreytek adapters today... Dropped them on to my test system. After wadding thru my Memory of how to setup mISDN, I had it up and running within about 2 hours. Both of them operating in ptmp with no echo cancel turned on at this stage. Seems to be happy. Will drop them onto my local production box next week and see how we go :D Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Sunday, 12 March 2006 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe Being USB 1.1 is not a problem - there is more than enough bandwidth for a BRI in USB. The handsets used in the BRI install are Snom 360's with firmware 5.3 and internal users have complained of slight echo, however I believe this is more to do with the Snoms than the Draytek adapters. For faxing use we have installed a Grandstream ATA 286. I haven't had any feedback yet regarding problems or success with faxing for this customer. I would have expected to hear of any problems faxing by now but I will try to follow it up, however as long as the latency is consistent (ie minimal jitter in the USB stack) it shouldn't cause any problems for fax. At work in our own office we have two SNOM 360's and people with them also complain of slight echo. (We are using TE110p PRI for PSTN). The rest of our office use a combination of Sipura 841, Cisco 7960 and Grandstream BT101 and there are no echo complaints with any of these non Snom handsets, so at this point it doesn't appear that these BRI adapters have echo problems. Craig - Original Message - From: "James Harper" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, March 12, 2006 7:35 AM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe I have ordered one (for $71 from the supplier you mentioned, although I have since found another supplier who appears to have them for $55!!!) and will run whatever testing I can. Someone from Cologne has commented that because it us a USB device, there may be some latency issues (which will amplify any echo problems) and I suspect that faxing may also suffer a bit. They are also only USB1.1, but I'm not sure if that's a problem. Have you tested faxing? Even if faxing doesn't work well enough to be useful because of the delays, I think this is a very nice solution to my problem (lack of BRI hardware in AU). Thanks again for bringing it to my attention!!! James > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Craig Guy > Sent: Friday, 10 March 2006 14:35 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe > > We didn't ask specifically for new ones. I believe the old ones went out > of > stock a long time ago. We ordered four at once and they all came with the > HFC chipset. > > Craig > > - Original Message - > From: "James Harper" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Friday, March 10, 2006 8:38 AM > Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe > > > > Note that > > it is only the currently available minivigors that have the HFCS-USB > > chipset, older ones on the secondhand market and eBay most likely use > a > > Winbond chipset. > > Is there any chance that they would sell me an old one? Do I need to ask > specifically that they supply the HFC one? > > Thanks > > James > ___ > --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 > > ___ > --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 ___ --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 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.1/279 - Release Date: 10/03/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.1/279 - Release Date: 10/03/2006 ___ --Bandwidth and Colocation provided by Easynews.com
RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe
I have just ordered a couple of them myself for a side project(like I don't already have enough to do!!!) Thanks for the heads up Craig Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 10 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe > Note that > it is only the currently available minivigors that have the HFCS-USB > chipset, older ones on the secondhand market and eBay most likely use a > Winbond chipset. Is there any chance that they would sell me an old one? Do I need to ask specifically that they supply the HFC one? Thanks James ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.1/278 - Release Date: 9/03/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.2.1/278 - Release Date: 9/03/2006 ___ --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] MultiBRI in Australia - found one - maybe
Best of luck :-D
I would be interested in your progress on this.
I am having very little problem in convincing ppl to upgrade their multiple
BRI cricuits for a single pri. The cost difference between a te110 (or a
Sangoma A101) MORE than covers the difference from the customer stand point,
especially once you are up to 3 ISDN-2 Interfaces.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Harper
Sent: Thursday, 9 March 2006 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe
I have received the card.
It comes with some closed source capi drivers, which I haven't tried as I
don't believe that is in acceptable solution anyway.
I had a look at hacking qozap to make it work, but haven't gone there at the
moment. What I'm looking at now is visdn. 0.14 doesn't even want to compile
against 2.6.15, but the latest development snapshot does, and after I added
in the correct PCI ID's, it detects the card.
I have no idea if the development vISDN HFC-4S drivers are even in a
workable state, but they do detect L1 status, and asterisk is able to detect
an incoming call but won't answer it.
The card itself is the 'Saphir III ML PCI'. Older versions of it used
another chipset ('Infineon' I think), but this newer one definitely uses the
HFC-4S chipset, and is definitely detected as such by the vISDN driver.
The only supplier I have found in Australia for it is
http://www.voipnow.com.au/, and they are the ones who have supplied the one
I am testing. On their web site, the picture is of the old version with 4
large chips on it, but the new one is pictured at
http://hstnet.de/english/index.asp.
I'll follow up if I have any further success, or if I give up.
James
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of David Hindmarsh
> Sent: Sunday, 5 March 2006 22:37
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one -
maybe
>
> Hi James,
>
> I am definitely interested in the card and also in the results of your
> testing.
>
> Regards,
>
> David
>
>
> LEXNET PTY LTD
> [e] [EMAIL PROTECTED]
> [m] 0411 172 667
> Mail: PO Box R1180
> Royal Exchange, Sydney NSW 1225
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of James
> > Harper
> > Sent: Saturday, 4 March 2006 12:03
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] MultiBRI in Australia - found one - maybe
> >
> > I may have found a source of an A-Ticked HFC 4BRI PCI adapter in
> > Australia, and will be testing one next week if all goes well. I
> > don't want to post the details of the reseller online unless invited
> > to do so, so if nobody replies and says they are interested then I
> > won't :)
> >
> > I'll follow up once I've tested it.
> >
> > Let me know if you want the details.
> >
> > James
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > Internal Virus Database is out-of-date.
> > Checked by AVG Free Edition.
> > Version: 7.1.375 / Virus Database: 267.15.11/264 - Release
> > Date: 17/02/2006
> >
> >
>
> --
> Internal Virus Database is out-of-date.
> Checked by AVG Free Edition.
> Version: 7.1.375 / Virus Database: 267.15.11/264 - Release Date:
> 17/02/2006
>
>
> ___
> --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
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RE: [Asterisk-Users] asterisk in real estate developments
We are working with a few Developers, but asterisk is only one part of the solutionbut we are using it for the telephony side of things, combined with Channel banks etc...etc..etc.. The Biggest Bugbear is billing. We are also rolling out and maintaining a GEPON structure...so everything travels over FTTH. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Wednesday, 14 December 2005 11:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] asterisk in real estate developments > I was wondering if anyone has used asterisk in a real estate > development project. I know someone that is developing a ~400 home > project and thought asterisk might be a possible alternative to the > phone company and a way to offer more service to buyers. How about deploying asterisk to support the contractor responsible for the construction of these sites? Instead of developers (who are often on-site for 6 months plus) relying purely on cellphones or asking the ILEC to install a load of phone lines for them, stick an asterisk server in their site office linked to a net connection, shove a load of cordless phones on a channel bank at convenient points around the site and contractors are never far from a phone. This is something we're hopefully doing for a property developer in the new year. It'll be interesting to see how well it all works out. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.13/199 - Release Date: 13/12/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.13/199 - Release Date: 13/12/2005 ___ --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] call center dial plan
Queues are your friend. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Queue From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ZlotySent: Friday, 2 December 2005 11:04 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] call center dial plan Hello How to write dialplan which will be doing something like this: I want to divide sip clients(consultants) into groups, And when call is incoming for example for number 6604, it will be redirected to first free random choose sip client from group 6604. Best regards Robert --No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.362 / Virus Database: 267.13.10/190 - Release Date: 1/12/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.10/190 - Release Date: 1/12/2005 ___ --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] Asterisk and Zaptel Versions Command?
Asterisk version mm*CLI> show version Asterisk 1.0.9 built by [EMAIL PROTECTED] on a i686 running Linux Zaptel, I am not sure but If you have built from CVS, the version info should be in the .version file in the src directory Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, 1 November 2005 3:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk and Zaptel Versions Command? On Mon, Oct 31, 2005 at 10:48:44AM -0800, Bart Fisher wrote: > Is there a command line for discovery of Asterisk and Zaptel Versions? I'm not aware of any way to tell the version of the zaptel kernel module. But then again, on my system there is the indirect way of 'dpkg -l zaptel-modules\*' -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.6/152 - Release Date: 31/10/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.6/152 - Release Date: 31/10/2005 ___ --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
RE: [Asterisk-Users] 2 box single Asterisk
Brave is the person that wants to use 3 Fritz cards in one box Go with the Jurgens 8 bri or 2 quad Brior bri-e1 chan bank... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Tuesday, 13 September 2005 6:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 2 box single Asterisk Here's my suggestion. Do a dialplan thing where when all trunks on boxA are busy, they are sent via IAX to boxB which sends them out via the ISDN trunks... this way boxA will be your primary box and boxB is your "spare" box that takes over if everything else is busy... On Tuesday 13 September 2005 10:00, Asterisk Sales wrote: > hello list, > i need to setup an asterisk system with 5 ISDN trunks. i found C4 > cards but they are very expensive. i found that if i use 5 AVM Fritz! > cards it would be very cheap. i want to use 2 boxes. 3 in boxA +2 in boxB =5 isdn. > and i want, this two boxs to work as a single box so that one box can > share ISDN hardware from other box. this system will be serving a call center. > currenly we are using a panasonic PBX system but it is driving us crazy. > we want to keep the existing pbx setup and add asterisk with it to > handle the call center operations. > we also need to communicate with pbx users from Asterisk. > our pbx has 6 analog trunks. so we can use TDM400P please help how > can i solve this situation will low cost and performance. > best regards > shaon ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.23/99 - Release Date: 12/09/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.24/101 - Release Date: 13/09/2005 ___ --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
RE: [Asterisk-Users] ParkAndAnnounce - No Disconnect
Quick and dirty way would be to then dump you into DISA and then retrieve the call from the parking lot Just a thought Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Edwards Sent: Sunday, 14 August 2005 10:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ParkAndAnnounce - No Disconnect Regarding my earlier email (for some reason, I don't get my own emails from the list), I looked at the code and although I'm no programmer, I see that this is meant to hangup after the announcement. If I comment that line out, the call remains on the line, but I'm in limbo. I tried to add a chan->priority = 101 in place of the hangup and get this on the cli: = Spawn extension (locator, s, 101) exited non-zero on 'Parked/SIP/3105989483-f5f1' So I'm not sure what happened there, but neither the parked caller or myself are affected. I'll look around at other code and see if I can figure out how to transfer the called party (me) into a context that can "Press 1 to accept the call" or some such thing. Maybe I'll even figure out how to pull the caller out of the parking lot and send them to voicemail if I choose to "Reject the Call" (instead of them having to wait for the timeout) Kris ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.8/71 - Release Date: 12/08/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.9/72 - Release Date: 14/08/2005 ___ 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] Detecting hangup - TDM400P / X100P
Have a look at the indications.conf file Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Davidson Sent: Tuesday, 9 August 2005 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Detecting hangup - TDM400P / X100P I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to Australia and input it into the Sipura. How do I input these tones into the TDM400P as being a hangup? TIA, tony Zero Effort Networking Pty Ltd ABN 38 082 434 446 PO Box 6045 Blacktown NSW 2148 www.zeroeffortnetworking.com.au [EMAIL PROTECTED] Tel: (02) 9676 3541 Fax: (02) 8569 2012 Message from: [EMAIL PROTECTED] Message to: asterisk-users@lists.digium.com Attached files: 0 This message contains confidential information and is intended for asterisk-users@lists.digium.com If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.2/65 - Release Date: 7/08/2005 ___ 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] Fritz PCI card in ptp mode with chan_misdn
Eric, I have a setup running chan_misdn in au with 3 fritz cards... I could never get Capi running happily and needing to support DID's didn't help much either! We are still having intermittent Echo Problems, but they are running in ptp mode happily. We are swinging away from this solution due to the high cost of A-Ticked Fritz Cards here and the Diva 4 Port Cards aren't much better. Once you hit 3 Onramp2 services it is almost more cost effective to put in a PRI-10 and use the TE110p Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop Sent: Wednesday, 27 July 2005 9:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fritz PCI card in ptp mode with chan_misdn Craig, You obviously have has experience with chan_mISDN in AU and the Fritz. Have you tried chan_capi? I am currently using a Fritz with chan_capi in AU and am not entirely happy with it. Is chan_mISDN any better? On 7/27/05, Craig Guy <[EMAIL PROTECTED]> wrote: > The mISDN Fritz! driver supports PTP mode. In your startup script > where you load the mISDN drivers call the fritz driver thusly: > > modprobe avmfritz protocol=34 > > Bit 5 sets PTP mode, bits 3-0 set the D-channel protocol ID (set bit > one for DSS1). > > Craig > > - Original Message - > From: "Michiel van Baak" <[EMAIL PROTECTED]> > To: > Sent: Wednesday, July 27, 2005 12:35 AM > Subject: Re: [Asterisk-Users] Fritz PCI card in ptp mode with > chan_misdn > > > > On 18:21, Mon 25 Jul 05, Johann Steinwendtner wrote: > > > Hello ! > > > > > > I would like to get working a Fritz PCI card using chan_misdn > > > operating in ptp mode. > > > > As far as I know the fritz cards do not support ptp mode. > > We tried all the possible config file options with chan_capi and in > > the end we trashed them and installed a junghanns QuadBRI. > > > > If you get it working in ptp mode, please tell me how you did it. > > -- > > Michiel van Baak > > http://michiel.vanbaak.info > > [EMAIL PROTECTED] > > GnuPG key: > > http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > > > "Why is it drug addicts and computer afficionados are both called users?" > > ___ > > 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 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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ___ 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] Extension Problems
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremi Bergman Sent: Friday, 8 July 2005 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Problems The extensions I've created in AAH, when dialed, always go straight to voicemail. I may be missing a step... I'm simply adding it in the "Extensions" part of AAH. I can dial out with my extension, and recieve the voicemail notification, so I know i'm logged in, or so I thought... This is SIP 210 logging in and 220 making a call to 210 <--SNIP --> Looks Like an Authentication Issue to me Chack the Username and password on the sip device and AAH Dave -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date: 6/07/2005 ___ 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] Asterisk@Home Ver 1.2 Whats new?
Hmmm..maybe the change log will reveal something... Notes: [EMAIL PROTECTED] - Taking calls in an hour --- version 1.2 - 06/29/05 Features: - * Asterisk 1.0.8 * Flash Operator Panel 0.21 * Festival Speech Engine version 1.95 * weather agi scripts * wakeup calls * Integrated WebMeetMe GUI * AMP-1.10.008 * CentOS 3.5 * SugarCRM with Cisco XML Services interface + Click to Dial * Music On Hold (mpg123) * Fax support (spanDSP) * xPL support * Digium card auto-config -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Shaw Sent: Thursday, 30 June 2005 10:46 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.7/34 - Release Date: 29/06/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.7/34 - Release Date: 29/06/2005 ___ 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] AMP/A@H (asterisk at home) custom incoming routing
You would be better using extensions_custom only because of the fact that when you restart ampportal, it will overwrite extensions_additional with what ever it has stored in the Database. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Wednesday, 29 June 2005 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AMP/[EMAIL PROTECTED] (asterisk at home) custom incoming routing Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or [EMAIL PROTECTED], there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs, which is all fine and good, unless you are using a plain old copper line that doesn't support DID. Anyhow, I have figured out how to make a call that comes in on a specific ZAP channel ring at a specific extension (not that it was brain surgery). I'm not certain if it would be better to use the file extensions_custom.conf instead of extensions_additional.conf, does anyone know? I have an [EMAIL PROTECTED] box with an unused TDM11P card in it at home in my basement. The [EMAIL PROTECTED] box normally handles incoming calls for my small business, but I wanted to plug my home phone line into the FXO port, and all of my phones into the FXS port (They're cordless, so no worries about ringer equivalence, etc.). That way I can route outgoing calls over VOIP, but my incoming calls will still ring my home phones. The hitch with [EMAIL PROTECTED] was that AMP doesn't allow you to differentiate between incoming calls from Broadvoice (or wherever) and incoming calls on the FXO port. I wanted incoming calls on my home line to ring to ring the extension associated with the FXS port on my TDM card. All other incoming calls should still follow whatever I set up in AMP, since that is how I control where my incoming business calls go. Here's what I did. I know that you aren't supposed to use the extension "_." but that is the only way I got it to work. Please let me know if there is a better way. 1.) Edit /etc/asterisk/zapata-channels.conf and change the context for your incoming port to something new. I used "tdm-in". 2.) Edit pico /etc/asterisk/extensions_additional.conf and add this at the bottom: [tdm-in] exten => _.,1,Goto(ext-local,200,1); 3.) If you haven't already, add the ZAP channel as a trunk in AMP so you can make outgoing calls on this channel. That's it, Tom ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.6/33 - Release Date: 28/06/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.6/33 - Release Date: 28/06/2005 ___ 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] Echo Issues
HI Callum, I am going thought a similar thing here with a site that I setup about 6 weeks ago... There doesn't seem to be any Pattern to it at this stage. I am *trying* to get the end users to keep a log of calls with echo to see if it is a specific channel Problem, Inbound/outbound etc. If I come up with anything more useful, I will let you know. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: Wednesday, 22 June 2005 11:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo Issues Hi all, We have just installed a * server (CVS head) with a TE110P card and a IDSN20 line, we are using the GXP-2000 handsets running the latest firmware (.9). Some of the calls we are receiving have echo at the our end (we can hear ourselves speak). We have a traditional ISDN telephone system here as well and when I make a call from one of the handsets asterisk answers, directs the call to one of our representatives and the calls are completely clear, however when we receive calls from customers at home, we often hear a lot of echo (not all the time, just sometimes). We have echo cancellation turned on and aren't really sure how we can get rid of this. Can anyone offer any suggestions ? Thanks, Callum ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.10/25 - Release Date: 21/06/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.10/25 - Release Date: 21/06/2005 ___ 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] gxp-2000 tftp cfg
If you download the "configuration tool" which I couldn't get working on my systemthere is a cfg template in there for 1.0.1.8 Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Wednesday, 8 June 2005 7:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] gxp-2000 tftp cfg On Tue, 7 Jun 2005, marek cervenka wrote: > can you someone post tftp template for gxp-2000? > like > http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandst > ream_Configuration_File_Template_1.0.6.x.txt I think it will be released with the 1.0.1.9 firmware. You may be able to get it by asking their support for it. YMMW. Peter __ -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.5 - Release Date: 7/06/2005 ___ 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] mISDN + chan_misdn.so + winbond issue
Update to at least chan_misdn-0.1.0 .. I am using snapshot from 11.05.05 without too many issues. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michel Koenen Sent: Monday, 6 June 2005 6:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] mISDN + chan_misdn.so + winbond issue Hi all, Does anybody of you have the winbond w6692 working with the mISDN/chan_misdn.so? When loading chan_misdn.so from Asterisk, I get a "No lower Id port:1" error. The /var/log/messages file says: "MISDN free_device: entitylist not empty" I'm using Linux 2.6.11.11 + mISDN-CVS-2005-05-01 + Asterisk 1.0.7 + Zaptel 1.0.7 chan_misdn build from chan_misdn-beta-0.0.3-rc6 and against mISDNuser-CVS-2004-08-29. The /dev/mISDN node was also created. I'm loading the kernel modules this way: modprobe zaptel modprobe ztdummy modprobe mISDN_core modprobe mISDN_l1 modprobe mISDN_l2 modprobe l3udss1 modprobe mISDN_dsp modprobe w6692pci protocol=2 layermask=1 Then I start asterisk: asterisk -c -vv -dd When loading chan_misdn.so , Asterisk complains and exits after the last error line below " [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 No lower Id port:1 init_stack: No such file or directory " Contents of the /var/log/messages for all above commands: Jun 5 20:25:20 pbx kernel: Zapata Telephony Interface Registered on major 196 Jun 5 20:25:25 pbx kernel: Registered tone zone 0 (United States / North America) Jun 5 20:25:48 pbx kernel: Modular ISDN Stack core $Revision: 1.25 $ Jun 5 20:25:53 pbx kernel: ISDN L1 driver version 1.11 Jun 5 20:25:56 pbx kernel: ISDN L2 driver version 1.20 Jun 5 20:26:02 pbx kernel: mISDN: DSS1 Rev. 1.29 Jun 5 20:26:07 pbx kernel: mISDN_dsp: Audio DSP Rev. 1.10 (debug=0x0) Jun 5 20:26:20 pbx kernel: Winbond W6692 PCI driver Rev. 1.13 Jun 5 20:26:21 pbx kernel: PCI: Found IRQ 9 for device :00:0f.0 Jun 5 20:26:21 pbx kernel: mISDN_w6692: found adapter Winbond W6692 at :00:0f.0 Jun 5 20:26:21 pbx kernel: W6692: Winbond W6692 version (0): W6692 V00 Jun 5 20:26:21 pbx kernel: w6692: IRQ 9 count 4 Jun 5 20:26:21 pbx kernel: w6692 1 cards installed Jun 5 20:26:34 pbx kernel: MISDN free_device: entitylist not empty Am I using wrong or incompatible source versions or is this a bug or am I doing something wrong? Btw the misdn.conf contains: [general] language=en immediate=no debug=0 [mycard] context=incoming ports=1,2 msns=72 Using ports=1 or ports=2 or changing msns gives the same problems.. When you have a working configuration, I am curious which source versions of needed packages you have used. Thank you in advance for your response. Best regards, Michel Koenen ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.2 - Release Date: 4/06/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.2 - Release Date: 4/06/2005 ___ 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] Sipura 3000 dialing "noise"
Have you updated with the lastest firmware..
It now does an on-hook forward to asterisk
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
Sent: Tuesday, 31 May 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sipura 3000 dialing "noise"
Hi all,
We have several sipura 3000's working well for outbound calls, however the
issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately and
then proceeds with the call "in band" therefore sending dialing sounds back
to the caller. Other SIP gateways we have notably the Vegastream and others
do not do a SIP answer until the call is successfully connected to the
called party.
Any ideas?
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RE: [Asterisk-Users] International Caller ID?
Anytime I receive a landline to anything over here in AUS, it comes up as Overseas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Malcolm-Smith Sent: Friday, 27 May 2005 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] International Caller ID? Rod Bacon wrote: > We have antiquated caller ID schemes here in Australia. We barely > support numbers from other local carriers, let alone OS ones. > Certainly no names either. When dialing out thru voipjet, I can put anything I like and it will come thru to my mobiles in New Zealand just fine (on both networks) - However calls to landlines just come up as on the caller ID as they put that for any international call. -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 ___ 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] chan_misdn problem
Can you post the output of your asterisk log file and your initd script for starting mISDN. What versions of chan_misdn, ,mISDN and mISDNuser are you using. Also check to see that /dev/mISDN exists. Dave. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of me me Sent: Thursday, 26 May 2005 9:18 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] chan_misdn problem I've installed asterisk 1.0.7 with linux kernel 2.6.3 (patched for mISDN). I Compile mISDNuser and loaded de modules (hfcmulti, mISDNdsp) for my BN8S0 beronet card. I have installed chan_misdn-beta-0.0.3rc4 with no problems. I have configured my misdn.conf as follows: [general] context=default language=de debug=0 immediate=no hold_allowed=yes [octoBRI] ports=1,8,2,7,3,6,4,5 context=incoming msns=* when I start asterisk with asterisk -vvvc I get the following message and then asterisk dies: [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) cannot request MGR_NEWENTITY from mISDN: Success Ouch ... error while writing audio data: : Broken pipe Warning, flexible rate not heavily tested! Can anyone help me?? Thanks. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 ___ 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] LOOKING TO HIRE
And this has driffted so far off topic, I don't understand why you all don't just go outside and duke it out in the carpark. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, 20 May 2005 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] LOOKING TO HIRE On Fri, 20 May 2005 02:23:47 +0400, Jean-Michel Hiver wrote: >Preston Garrison wrote: > >> Java compiles into bitcode, that a bitcode interpreter runs. Its >> kinda in a state of flux between the two :) > >Well, first it's bytecode. > >Second, bytecode is no different from binaries since a bytecode >processor perfectly doable. > >Third, Perl, Python and PHP can also be compiled down to bytecode. > >Fourth, Scheme and Lisp are very similar to OCaml (LISP family of >languages). Scheme and Lisp are interpreted, while OCaml is compiled. >It does not change the completeness of each language... > >Ergo, the assertion Good Programmer = Compiled Languages is *pure bull*. Good programmer = assembly language! Just kidding ;-) Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.13 - Release Date: 19/05/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.13 - Release Date: 19/05/2005 ___ 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] Grandstream GXP2000 firmware update
Yes...no Problems...used TFTP Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Thursday, 12 May 2005 7:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Grandstream GXP2000 firmware update I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream GXP2000 1.0.0.3" [Wed May 11 16:47:17 2005] [error] [client 201.133.125.152] File does not exist: /usr/local/apache/htdocs/voip/firmware/ring2.bin 201.133.125.152 - - [11/May/2005:16:47:17 -0500] "GET /firmware/ring2.bin HTTP/1.0" 404 289 "-" "Grandstream GXP2000 1.0.0.3" [Wed May 11 16:47:18 2005] [error] [client 201.133.125.152] File does not exist: /usr/local/apache/htdocs/voip/firmware/ring3.bin 201.133.125.152 - - [11/May/2005:16:47:18 -0500] "GET /firmware/ring3.bin HTTP/1.0" 404 289 "-" "Grandstream GXP2000 1.0.0.3" [Wed May 11 16:47:19 2005] [error] [client 201.133.125.152] File does not exist: /usr/local/apache/htdocs/voip/firmware/cfg000b8200 201.133.125.152 - - [11/May/2005:16:47:19 -0500] "GET /firmware/cfg000b8200 HTTP/1.0" 404 295 "-" "Grandstream GXP2000 1.0.0.3" [Wed May 11 16:47:21 2005] [error] [client 201.133.125.152] File does not exist: /usr/local/apache/htdocs/voip/firmware/cfg.txt 201.133.125.152 - - [11/May/2005:16:47:21 -0500] "GET /firmware/cfg.txt HTTP/1.0" 404 287 "-" "Grandstream GXP2000 1.0.0.3" ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 10/05/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 10/05/2005 ___ 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] Connecting 2 * Together-Pulling hair out
I Had the same Problem as you did... I used the following from the list as a template and Setup up my dial Plan Accordingly... http://lists.digium.com/pipermail/asterisk-users/2004-September/062564.html Hope it helps. Dave Chris wrote: >I haven't gotten to keys yet. >The documentation out there doesn't seem to be very good. > >Chris > > >- Original Message - >From: "Tim Pushor" <[EMAIL PROTECTED]> >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > >Sent: Thursday, May 05, 2005 4:06 PM >Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out > > > > >>Personally, if I owned both boxes and had full control of the dialplan >>on both, I'd stay away from passwords. (but be careful what I say, I'm >>a >>hack) >> >>I have a bunch of boxes connected together via IAX and authenticating >>via RSA. The entries in iax.conf are simple, and dialing across the >>connection is simple (no passwords in the dialplan) (thanks again Rich >>for taking the time). >> >>Tim >> >>Here is a sample of iax.conf entries on machine a: >> >>[machineb] >>type=user >>host=machineb.internal.net >>auth=rsa >>inkeys=machineb >>username=machineb >>context=inbound >> >>[machineb] >>type=peer >>host=machineb.internal.net >>auth=rsa >>outkey=machinea >>username=machinea >> >>And an example dialplan entry to dial an extention on machineb (in the >>inbound context): >> >>exten => 333,1,Dial(IAX2/machineb/333) >> >>And on machinea, the opposite of machineb: >> >>[machinea] >>type=user >>host=machinea.internal.net >>auth=rsa >>inkeys=machinea >>username=machinea >>context=inbound >> >>[machinea] >>type=peer >>host=machinea.internal.net >>auth=rsa >>outkey=machineb >>username=machineb >> >>To generate the keys: >> >>on machinea: >> >>astgenkey -n machinea >>mv machinea.* /var/lib/asterisk/keys >> >>copy machinea.pub to machineb's /var/lib/asterisk/keys >> >>on machineb: >> >>astgenkey -n machineb >>mv machineb.* /var/lib/asterisk/keys >> >>copy machineb.pub to machinea's /var/lib/asterisk/keys >> >> >>Chris wrote: >> >> >> >>> I have something similar. Both of my servers are behind a firewall and NAT. You will need to allow UDP 4569 through the firewall for IAX2. If you have NAT you will need to redirect 4569 to the internal server. >>> >>> I would suggest using AMP and then looking at IAX_ADDITIONAL.CONF to see how it's done. You can modify the IAX.CONf because I don't believe AMP rewrites that file. >>> >>> I think the user and passwords are required. I would suggest using a strong password or someone may decide to make a few phone calls. After this you will need the routing in Extensions.conf to allow calls to be made on this trunk. >>> >>> Asterisk will handle the SIP > IAX.All my clients are SIP and they have no trouble going over a IAX trunk to other SIP devices on the other server. >>> >>>This is what my IAX_ADDITIONAL.CONF looks like >>> >>>SiteA - Dynamic IP >>>-- >>>[boxb-peer] >>>username=boxa-user >>>type=peer >>>trunk=yes >>>secret=mypassword >>>host=thehost.dyndns.org >>> >>>[boxb-user] >>>type=user >>>secret=mypassword2 >>>host=thehost.dyndns.org >>>context=from-internal >>> >>>--- >>>Site b - Static IP >>> >>> >>>[boxa-peer] >>>username=boxb-user >>>type=peer >>>trunk=yes >>>secret=mypassword2 >>>host=xxx.xxx.xxx.xxx >>> >>>[boxa-user] >>>type=user >>>secret=mypassword >>>host=xxx.xxx.xxx.xxx >>>context=from-internal >>> >>> >>>Regards, >>> >>>Chris >>> >>> >>>- Original Message - >>>From: "mr. barker" <[EMAIL PROTECTED]> >>>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" >>> >>>Sent: Thursday, May 05, 2005 1:58 PM >>>Subject: RE: [Asterisk-Users] Connecting 2 * Together-Pulling hair >>>out >>> >>> >>> >>> >>> >>> Yes trying to connect to boxes together. One sits outside the internal firewall and is on the inside. I am using AMP. However I can just put whatever I need in the custom.conf sections. The users agents are SIP .. can SIP call go over a IAX trunk ? if so great. To create the trunk do I need to use a users name and password ? or ? I need to have the *box that is behind the firewall to be able to place a call out through the *box that has a public ip. Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sent: Thursday, May 05, 2005 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out I am not sure what you are trying to do.I have created an IAX2 trunk between the servers over an internet connection. Then all you have to do is put in call routing on the trunks to forward the call to the right place. Are you using AMP or trying to do it manually. I found everything a little confusing as well, but it is simple now
RE: [Asterisk-Users] TE410P does not fit in motherboard
[Oooops] Corect me if I am wrong, but the TE410P is for 5v PCI Slots.. I think you need to be using the TE405P (3.3V PCI) [/ooops] Got that back-to-front Maybe I should Have had the Scrambled Eggs instead of my Brain... Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, 5 May 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE410P does not fit in motherboard I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and I'm just noticing that the TE410P does not fit in the PCI slot. It seems as if the little opening in the PCI is on the wrong side. Has anyone else seen this or is it just me and I'm too stupid to do something as basic as this? Thanks, Daniel ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.5 - Release Date: 4/05/2005 ___ 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] TE410P does not fit in motherboard
Corect me if I am wrong, but the TE410P is for 5v PCI Slots.. I think you need to be using the TE405P (3.3V PCI) Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, 5 May 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE410P does not fit in motherboard I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and I'm just noticing that the TE410P does not fit in the PCI slot. It seems as if the little opening in the PCI is on the wrong side. Has anyone else seen this or is it just me and I'm too stupid to do something as basic as this? Thanks, Daniel ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.5 - Release Date: 4/05/2005 ___ 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] Attended Transfer using wrong Context
Can you post the context for cytel-outgoing... >From what it sounds like..asterisk is picking the # as a blind transfer then 9 which means you are trying to transfer to an outside number an ddepending on your dial plan, that may not work. I do realise that you are trying to use attended transfer so maybe change the attended transfer sequence so that it doesn't use 9. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, 5 May 2005 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Attended Transfer using wrong Context Its set to 3 seconds. When I hit the "3" to being typing "3069" it is an immediate error. The problem is that asterisk uses the context of the #1 call to place the Attended Call; instead of using the context that the phone is registered to. -Matthew Noah Miller wrote: >> The phone's context is "cytel-internal". >> This allows us to hit "3XXX" to get someone on the inside. >> >> If you hit "9" at the beginning, you Goto() the "cytel-outgoing" >> context. >> >> So lets make a call..I'll dial 918005551212 (toll free directory). >> >> The 9 sends it to cytel-outgoing. Call is made. Bridged. I then hit >> #9 for >> attended transfer. >> >> Allison says "Transfer". I start to enter 3013. But right after I hit >> the first 3, it returns failed transfer: >> >> res_features.c:800 builtin_atxfer: Did not read data. >> >> Wtf? >> >> So I do it again; and again. I tried every number and they all >> returned the same error. >> >> But this time I press 93013 and the call goes out the >> "cytel-outgoing" context. >> >> ???!?? >> >> I'm lost. What is this thing doing? > > Being very bad. > > Just some ideas: > > What's the "transferdigittimeout" setting in features.conf? Maybe > it's not giving you enough time to really enter an extension. Also, > what happens when you change attended transfer to something other than > "#9" > > - Noah > > ___ > 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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.2 - Release Date: 2/05/2005 ___ 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] How do "take away" do not disturb from certainphones
Easy.. go to the web interface of the Handset you want to modify... go to admin login then select advanced go to the 'phone' tab, and under suplementary Services there is a whole list of things that you can enable and disable on the phones.. Including DND Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, 4 May 2005 4:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How do "take away" do not disturb from certainphones Hi, I'm wondering if anyone has an idea on how to disable "Do not disturb" for certain phones. I have several Sipura SPA-841s. They work fine for the application we are using them for. However, the menus are clutzy and sometimes employees put them on DND by accident and "forget" to take them off. I'd like to disable the DND function on those phones but I don't see anything simple on how to do that. Any ideas? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.2 - Release Date: 2/05/2005 ___ 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] missing first digit when dial extension / dtmfproblem ???
Wtihout seeing any conf files...it's a bit hard to say Are you sure you haven't don't something strage with your dial plans? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Friday, 29 April 2005 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] missing first digit when dial extension / dtmfproblem ??? I'm using dtmfmode=inband with Sipura=3000 when I dial an internal extension most of the time the first digit is missing and I get an invalid extension message. Could it be dtmf problem or SIP? -- #Joseph ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.4 - Release Date: 27/04/2005 ___ 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] RJ45 to RJ11?
I sit corrected Thinking of an Ericcson BP250 Config.. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan & Company, LLC Sent: Thursday, 28 April 2005 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RJ45 to RJ11? This is incorrect, David - Pins 4 and 5 are the correct pins, and are not reversed. David Phelan wrote: > I think you will find it is pin reversed. > So flip the RJ45 Over > > Dave > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Gregory > Junker > Sent: Thursday, 28 April 2005 4:29 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] RJ45 to RJ11? > > Connect the POTS pair to pins 4 and 5 in the RJ45, and you should be > fine (I say this not having looked at the TDM400 specs, but from the > perspective of standard wiring practice and the assumption that Mark et al followed same). > > Greg > > Paul Shiflet wrote: > > >>I just received my TDM400 card from digium with 2 fxo and 2 fxs >>interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS >>phones. How do i interface my POTS phones with this; can i just crimp >>an >>RJ45 connection on the end of the phone cord? >> >>Paul >> >> >>___ >>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 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 > > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.10.3 - Release Date: 25/04/2005 > > > ___ > 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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.4 - Release Date: 27/04/2005 ___ 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] RJ45 to RJ11?
I think you will find it is pin reversed. So flip the RJ45 Over Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Thursday, 28 April 2005 4:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RJ45 to RJ11? Connect the POTS pair to pins 4 and 5 in the RJ45, and you should be fine (I say this not having looked at the TDM400 specs, but from the perspective of standard wiring practice and the assumption that Mark et al followed same). Greg Paul Shiflet wrote: >I just received my TDM400 card from digium with 2 fxo and 2 fxs >interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS >phones. How do i interface my POTS phones with this; can i just crimp >an >RJ45 connection on the end of the phone cord? > >Paul > > >___ >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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.3 - Release Date: 25/04/2005 ___ 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] Turn off Music on Hold
In modules.conf change load => res_musiconhold.so to noload => res_musiconhold.so -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg Sent: Wednesday, 27 April 2005 6:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Turn off Music on Hold I'm getting these: Apr 26 12:59:02 NOTICE[14775]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 26 12:59:02 WARNING[14775]: res_musiconhold.c:205 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Apr 26 12:59:02 WARNING[14775]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player Since I don't have any music on hold, and don't want any, how can I turn MOH off all together? ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.3 - Release Date: 25/04/2005 ___ 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] Asterisk Restart after crash
After a crash of what?? Linux...asterisk?? Depends on how you have it setup If you start asterisk with safe_asterisk, then if asterisk crashes it will start again. If you run safe_asterisk from say...your rc.local then it will start when linux restarts. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith Sent: Friday, 22 April 2005 1:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk Restart after crash Does Asterisk restart itself if it crashes? If not is there a way to make linux do it? ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 21/04/2005 ___ 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] Citrix
Not to mention any additional latency that you could be introducing.. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, 19 April 2005 3:38 PM To: Javier Godinez; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Citrix You should get an award for coming up with this idea. What for? if you have a computer to connect to citirx, the you should use the computer and *NOT* citrix for the soft client. I don't think that citrix has better compression for sound than VoIP. On 4/18/05, Javier Godinez <[EMAIL PROTECTED]> wrote: > Has anyone out there found a VoIP client that is citrix compatible? I > am connecting to a virtual machine via citrix and want to launch a > citrix compatible soft phone to connect to another virtual machine > running asterisk. Does anyone have a similar setup out there? > > Thanks, Javier > ___ > 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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.16 - Release Date: 18/04/2005 ___ 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] Sipura SPA-841 and Asterisk 1.0.7 with chan_misdn
HI Everyone, I have run into a rather unusual Problem.. My Config as follows System 2.6.9 Kernel mISDN 0.0.3.RC6 AVM Fritz! X 3 chan_misdn-0.1.0 Asterisk CVS Stable. Handsets: Micronet SP5100 Micronet SP5001 ATA Sipura-841 (Latest FIrmware) When I Make Calls from the SPA to PSTN(or the reverse), at first calls go through clear. After the Second or third Call, we wind up with 4-7ms jitter. If I transfer the call to the Micronet(which doesn't seem to experience ANY difficulties), call is cleartransfer back to the SPAjitter again the Jitter is only heard on the SPA end...the PSTN end of the call is fine Calls from sip to sip present no issues, as with calls to IAX2 trunks. Has anyone else run into this difficulty, or at least point me in a direction to try and fault find this Much Thanx Dave ___ 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] Asterisk management portal
Try wwwadmin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D Schelin Sent: Tuesday, 12 April 2005 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk management portal Hi everyone, Why doesn't this work? I can't get in. Is it because I changed the root? User: admin Pass: password ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.6 - Release Date: 11/04/2005 ___ 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] UK ISDN with Asterisk
I have a woking system now with 3 fritz cards with DID running chan_misdn.. Take Capi out of the Picture all together and it works fine. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Saturday, 9 April 2005 7:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK ISDN with Asterisk This would be a good solution but be aware that at this time the Fritz! may not handle DID (specifically PTP mode). The AVM drivers will not support DID. The mISDN drivers and fritz! cards do seem to handle DID but chan_capi doesn't pass the call to Asterisk (although you can see the call coming in with capi debug enabled). You might be able to get DID and frtiz! working with a combination of mISDN drivers (Kernel 2.6.9, 2.6.10 won't detect the fritz! card) and chan_mISDN. Craig - Original Message - From: "Gavin Hamill" <[EMAIL PROTECTED]> To: Sent: Saturday, April 09, 2005 7:01 AM Subject: Re: [Asterisk-Users] UK ISDN with Asterisk > On Friday 08 April 2005 23:33, Henry Owens wrote: > > Hi all, > > > > > My question is: can Asterisk work well as a small office (8 extensions) > > PBX, with a mixture of analogue and IP phones, on an ISDN2e telephone > > line from BT? > > Sure, no problem at all.. > > Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a 'BT > Speedway ISDN' adapter - these seem to be the most cheap and supported of > low-end ISDN2 adapters. > > chan_capi will deal with things like both B-channels so you can happily > receive two calls on the same number, and deal with MSNs (Multiple Subscriber > Numbers) gracefully since these are more likely on UK ISDN2e service than > true DDIs. > > gdh > ___ > 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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 7/04/2005 ___ 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] Australia and SetCallerID
>From what I have read, it should work on PRI and BRI with DID I will be trying this setup in about 8 hours. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nate Kapi Sent: Thursday, 31 March 2005 4:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Australia and SetCallerID Simple questionIs it possible to use SetCallerID in Australia? Has anyone found a provider that allows it? Thanks! ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 29/03/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 29/03/2005 ___ 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] Kernel panic loading second fritz card
Hi Everyone, Long time reader, first time poster. FINALLY got my First AVM Fritz Card up and running under Centos 3.4 Installed the secondmodified the drivers etc as per the instructions found at the wiki System boots Modprobe capi all good modprobe fcpci all good modprobe f2pci the kernel then goes into Panic If Modprobe f2pci before fcpci the kernel still goes into panic. Config as follows CentOS 3.4 Kernel 2.4.21-27.0.2.EL fcpci - fcpci-suse8.2-03.11.02 chan_capi-0.3.5 _ David Phelan Blue Ridge Systems Ph:+61 7 3624 8777 _ -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 27/03/2005 ___ 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
