[Asterisk-Users] UK Patches for Asterisk 1.2
As I am trying to compile a fresh copy of the current svn release of Asterisk 1.2 for a UK system with a combination of X100 and TDM cards, can a kind soul email me the CLID patches for 1.2? Many thanks Vassilis ___ --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: Fw: Re: Re: [Asterisk-Users] Zaptel errors on Debian
On Fri, Dec 02, 2005 at 10:06:09PM -0800, Geo wrote: Message not sent Right, not the comma separator I still have 2 small questions: 1) I have the following warnings: WARNING[2943]: chan_zap.c:10916 setup_zap: Ignoring :callreturn WARNING[2943]: res_musiconhold.c:124 spawn_mp3: /usr/share/asterisk/mohwav is not a valid directory WARNING[2943]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player This is harmless, basically. The mp3 files that are distributed with Asterisk have a problematic license and are thus not included in the Debian package (and even removed from the source package). BTW: playing mp3 files for music-on-hold is a waste of CPU cycles, as you need to transcode and down-sample the files over and over again. Which is why I have integrated a wavplayer in our (Xorcom Rapid) 1.0 packages. 1.2 should support native music on hold but I didn't yet get to try it. Ofcourse, I have no more chan_zap.c nor res_musiconhold.c. Musiconhold=no in zapata still warnings CLI 2) How can I remove BRIstuffed ? I'm working on providing non-bristuffed packages. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] UK Patches for Asterisk 1.2
On Sun, Dec 04, 2005 at 08:14:07AM +, Vassilis Konstantinou wrote: As I am trying to compile a fresh copy of the current svn release of Asterisk 1.2 for a UK system with a combination of X100 and TDM cards, can a kind soul email me the CLID patches for 1.2? As announced before in this list: try http://www.lusyn.com/asterisk/patches.html However it is currently not included in my packages as it seems to conflict with current version of bristuff. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] [EMAIL PROTECTED] isdn
On Sat, Dec 03, 2005 at 09:28:57AM +0100, Karsten Wemheuer wrote: Hi, On Tue, November 29, 2005 13:50 Francesco Peeters wrote: BTW: BRIstuff is not included by default as it breaks PRI support. Asterisk is already set up to use zap, so that is easy... As far as I know, BRIstuff is not included for licencing reasons... That is: bristuff is fully free (GPL). Its author does not feel inclined to allow Digium to relicense those changes. Is it true, that PRI support and BRIstuff are now incompatible? Considering that one of the cards in bristuff (the driver name is cwain, which stands for Card Without An Interesting Name, honestly) is a PRI card, it can't be that incompatible. However libpri in bristuff is more and more a rewrite rather than a patch. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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
[Asterisk-Users] Sipura 3000 Disconnect Singnel
Hi all, I was testing the FXO system from sipura 3000 with asterisk PERL AGI. But when we hangup the FXO phone the channel is not disconnecting and the destination is continue ringing. even if we try to press the disconnect button for destionations after some seconds again it start to ringing. Is it problem from Sipura itself or i have to do some advance in asterisk configuration. You suggestion will be high appricated. -- Thank You, Code Lover ___ --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] UK Patches for Asterisk 1.2
Thanks Tzafrir, But those patches do not work with the current 1.2. The asterisk_uk patch fails on 3 accounts with chan_zap.c and callerid.h The zaptel patch seems to be ok. Vassilis At 09:00 04/12/2005, you wrote: On Sun, Dec 04, 2005 at 08:14:07AM +, Vassilis Konstantinou wrote: As I am trying to compile a fresh copy of the current svn release of Asterisk 1.2 for a UK system with a combination of X100 and TDM cards, can a kind soul email me the CLID patches for 1.2? As announced before in this list: try http://www.lusyn.com/asterisk/patches.html However it is currently not included in my packages as it seems to conflict with current version of bristuff. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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
Re: [Asterisk-Users] Iax2 connection failed
On 3 Dec 2005, at 21:41, chawki hammoud wrote:Hi:Thanks for your answer, i tried all possible codecsand the same result the call failed,my asteriskverison is 1.0 ,I asked callshopcompany "the voipprovider" about whats the reason of the failure of thecalls and he said he didnt know whats the problem andhe's all customers making succesful calls to their iaxserver without any problems.NOTICE: i make successful calls through sip to thesame voip provider. Can you send us the exact version of asterisk?what do you get back from asterisk -rv ?On my system I get :Asterisk 1.0.3, Copyright (C) 1999-2004 Digium.Written by Mark Spencer [EMAIL PROTECTED]=Connected to Asterisk 1.0.3 currently running on risk (pid = 11808)Verbosity was 0 and is now 1Tim. http://www.westhawk.co.uk/ ___ --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 queues, agents with DND status set.
there should be a way in agents.conf to autologoff agents after a while the do not answer the phone. l. In data Sat, 03 Dec 2005 23:48:05 +0100, Vladimir S. Blazhkun [EMAIL PROTECTED] ha scritto: -- Called 1101 -- Agent/1101 is ringing -- Got SIP response 480 Temporarily Unavailable back from x.x.x.x -- SIP/1101-9b08 is circuit-busy Is it possible to force logoff such agents? -- Assum est, versa et manduca. ___ --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] Iax2 connection failed
HI: i tried to write asterisk -rv on console but no such command message appears,but when i make show version it gives me this: Asterisk CVS-v1-0-08/22/05-18:56:48 built by [EMAIL PROTECTED] on a i686 running Linux. --- tim panton [EMAIL PROTECTED] wrote: On 3 Dec 2005, at 21:41, chawki hammoud wrote: Hi: Thanks for your answer, i tried all possible codecs and the same result the call failed,my asterisk verison is 1.0 ,I asked callshopcompany the voip provider about whats the reason of the failure of the calls and he said he didnt know whats the problem and he's all customers making succesful calls to their iax server without any problems. NOTICE: i make successful calls through sip to the same voip provider. Can you send us the exact version of asterisk? what do you get back from asterisk -rv ? On my system I get : Asterisk 1.0.3, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.0.3 currently running on risk (pid = 11808) Verbosity was 0 and is now 1 Tim. http://www.westhawk.co.uk/ ___ --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 __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --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
[Asterisk-Users] replace hard-phone if soft-phone is online
How can I configure asterisk to switch from my hardphone which is always up and online, as soon as I register with my notebook's softphone on the asterisk server? The target is to receive all calls destinated to my hardphone on my softphone when it's online. Any ideas? ___ --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] replace hard-phone if soft-phone is online
dima wrote:
How can I configure asterisk to switch from my hardphone which is
always up and online, as soon as I register with my notebook's
softphone on the asterisk server?
The target is to receive all calls destinated to my hardphone on my
softphone when it's online.
Any ideas?
I'm just a newbie, but I've been doing something like this :
[adialplan]
exten = _,1,Set(calling=${EXTEN:0:4})
exten = _,2,ChanIsAvail(SIP/[EMAIL PROTECTED])
exten = _,3,Dial(SIP/[EMAIL PROTECTED],15,tTrwW)
exten = _,4,Goto(_-${DIALSTATUS},1)
exten = _,103,Dial(Zap/1/${calling},15,tTwWr)
exten = _,104,Goto(_-${DIALSTATUS},1)
The idea is to use ChanIsAvail to see if the SIP is online and if so use
it, otherwise, call the POTS line.
JES
smime.p7s
Description: S/MIME Cryptographic Signature
___
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Re: [Asterisk-Users] UK Patches for Asterisk 1.2
Vassilis Konstantinou wrote: Thanks Tzafrir, But those patches do not work with the current 1.2. The asterisk_uk patch fails on 3 accounts with chan_zap.c and callerid.h The zaptel patch seems to be ok. Worked fine for me... Are you sure you used the 1.2.0 patch? Tony ___ --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
[Asterisk-Users] re: Help required on asterisk
hi all, some days back i mailed to group abt my error on asterisk. Now also i m getting same error : cannot find extension context 'from-sip' . I tried DEFAULT context also. but at that time errot remains same: cannot fined extension context 'default'. I think problem is that it is not recognising any context. well i want some suggesion from group. It will be helpful for me if u will send ur suggestions. Actually I want to implement VoIP gateway for my Project work. For that i choose Asterisk. I have certain questions :1) Now i m planning to choose Debian as an operating system. So what do u thinkhows asterisk support for debian. Is asterisk work well on debian? 2) Also if i will use digitnetworks X100P card for PSTN interface would i able toconnect with analog phone. Is only this hardware is sufficient for making gateway? 3) And last but not leastsince i m a newbie to asterisk n Debian how much time(approximately in days) it will take to develop VoIP gateway with asterisk ? sorry i m giving u all a lot of pain, but sir ur suggestions will be very much valuable for me. Thanks, waiting for ur reply Tejas Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Try Yahoo! Personals___ --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] Iax2 connection failed
On 4 Dec 2005, at 13:33, chawki hammoud wrote:HI:i tried to write "asterisk -rv" on console but "nosuch command" messageappears,but when i make "show version" it gives methis:Asterisk CVS-v1-0-08/22/05-18:56:48 built by[EMAIL PROTECTED] on a i686 running Linux.Weird. IAX should be fine with that version, but I suppose it wouldn't hurt to upgrade to the1.0.9 (or 10) final 1.0 release version.I don't know what the problem is, but as I said, the next move would be to runethereal to capture the packets and see what the cause code actually was.T. http://www.westhawk.co.uk/ ___ --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] re: Help required on asterisk
For starters, please provide a more descriptive subject for your message. Something like cannot find extension context 'from-sip' would be better than nothing. Also see my reply inline. On Sun, Dec 04, 2005 at 07:48:11AM -0800, Tejas Shah wrote: hi all, some days back i mailed to group abt my error on asterisk. So consider continuing that thread. Now also i m getting same error : cannot find extension context 'from-sip' . I tried DEFAULT context also. but at that time errot remains same: cannot fined extension context 'default'. I think problem is that it is not recognising any context. Could you please provide the relevant parts of your extensions.conf ? Do you connect from a SIP channel? What is the context of that channel? use 'sip show users' well i want some suggesion from group. It will be helpful for me if u will send ur suggestions. Actually I want to implement VoIP gateway for my Project work. For that i choose Asterisk. I have certain questions : 1) Now i m planning to choose Debian as an operating system. So what do u think hows asterisk support for debian. Is asterisk work well on debian? Sure :-) I can also recommend http://xorcom.com/rapid, which is a Sarge with a better Asterisk. But then again I'm not an impartial observer. 2) Also if i will use digitnetworks X100P card for PSTN interface would i able to connect with analog phone. Is only this hardware is sufficient for making gateway? Basically, yes. There are some potential problems with X100P/X101P cards (e.g: echo problems). But then again, they're much cheaper than the alternatives... 3) And last but not leastsince i m a newbie to asterisk n Debian how much time (approximately in days) it will take to develop VoIP gateway with asterisk ? Not that much. See the above link for a shortcut (also note the license, which is GPL) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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
[Asterisk-Users] ISDN 2e Cards
I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK but both seem to have drawbacks/advantages. I need to build a new Asterisk box for my tiny business (1 x ISDN2e from BT and 1 x IAX link from Gradwell) Is anyone prepared to go out on a limb and say which card they prefer and why? TIA Simon ___ --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] ISDN 2e Cards
Simon Faulkner wrote: I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK but both seem to have drawbacks/advantages. I need to build a new Asterisk box for my tiny business (1 x ISDN2e from BT and 1 x IAX link from Gradwell) Is anyone prepared to go out on a limb and say which card they prefer and why? TIA Simon ___ --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 BT SpeedWay PCI cards are ideal. Work perfectly with chan_capi-cm for ISDN2e and Business Highway here in the UK. They are basically AVM Fritz cards badged by BT. I have a stock of brand new ones if you need, or alternatively they are often advertised on the auction sites (new and used). Faris. ___ --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
[Asterisk-Users] Why does musiconhold.conf changes require a reboot?
Hi, Why do changes to musiconhold.conf require a reboot. Also if I put mp3's into the /var/lib/asterisk/mohmp3 directory will the be played if I use the -r option? Using Asterisk 1.2 and have run the make config in the /usr/src/asterisk-addons directory. Thanks ___ --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
[Asterisk-Users] New to [EMAIL PROTECTED]
I'm thinking of Installing [EMAIL PROTECTED] on my PC, to contol and route calls to each of my children's computer via SoftPhone X-lite. I downloaded the ISO image, and familiar with the process to install Asterisk now. However I don't have a digium modem. Can I use any regular phone modem for incoming line on the Asterisk server? ---Dakota the Newbie ___ --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
[Asterisk-Users] RE: how to remove asterisk 1.2 from Red Hat 9
hi all,Can anyone tell me how i can remove (uninstall) asterisk 1.2 from Red Hat 9. Also pls tell me which version of asterisk is most suitable for making VoIP gateway on Red Hat 9.Thankstejas Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals___ --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] ISDN 2e Cards
Faris Raouf wrote: Simon Faulkner wrote: I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK but both seem to have drawbacks/advantages. I need to build a new Asterisk box for my tiny business (1 x ISDN2e from BT and 1 x IAX link from Gradwell) Is anyone prepared to go out on a limb and say which card they prefer and why? TIA Simon ___ --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 BT SpeedWay PCI cards are ideal. Work perfectly with chan_capi-cm for ISDN2e and Business Highway here in the UK. They are basically AVM Fritz cards badged by BT. I have a stock of brand new ones if you need, or alternatively they are often advertised on the auction sites (new and used). I'd second that. For a single ISDN2e connection the AVM Fritz card is really hard to beat/ jd -- John Daragon [EMAIL PROTECTED] 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 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] ISDN 2e Cards
John Daragon wrote: I'd second that. For a single ISDN2e connection the AVM Fritz card is really hard to beat/ Yeah, single is the key word there. I have 2x ISDN2 (OnRamp2 in Australia) and the AVM Fritz cards are a nightmare. Replaced the two cards with an Eicon Diva V-4BRI (so I have two extra ports if necessary) and my Asterisk box is just incredible now: Almost zero echo across the board and much lower processor utilisation. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 2 6233 0607 Fitzroy, VIC F: +61 (0) 2 6233 0696 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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] Failover Registration
You can simply put then in order in your dial plan: exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED] exten = _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED] and so on. if carrier1 returns an error, * will dial out using carrier2. SciptHead On 12/2/05, Max Clark [EMAIL PROTECTED] wrote: Hi all,I would like to have two asterisk servers in a cluster. From what Iunderstand using a mysql database I can store all of my peer/userinformation in the db and share this between servers. I can then take my polycom phone and register it to both of the asterisk servers at thesame time - so if one were to go offline traffic would be redirected tothe second.This works in theory for the end user - but how do I provide redundancy with my upstream providers? I.e. how do I fail over my registration toan upstream sip provider?Thanks in advance,Max-- Max Clark max [at] clarksys.com http://www.clarksys.com___--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
Re: [Asterisk-Users] Iax2 connection failed
Hi: Sorry,but i dont know what ethereal is,and for my asterisk version the iax is good on it because i made a lot of succesful iax connections with many voip providers like sixtel,voipjet... --- tim panton [EMAIL PROTECTED] wrote: On 4 Dec 2005, at 13:33, chawki hammoud wrote: HI: i tried to write asterisk -rv on console but no such command message appears,but when i make show version it gives me this: Asterisk CVS-v1-0-08/22/05-18:56:48 built by [EMAIL PROTECTED] on a i686 running Linux. Weird. IAX should be fine with that version, but I suppose it wouldn't hurt to upgrade to the 1.0.9 (or 10) final 1.0 release version. I don't know what the problem is, but as I said, the next move would be to run ethereal to capture the packets and see what the cause code actually was. T. http://www.westhawk.co.uk/ ___ --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 __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --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] Failover Registration
With this setup Asterisk will also dial out using carrier2 even if the call via carrier1 does NOT fail. It will dial out via carrier2 if the number is busy, disconnected, answered then ended, etc. This is BAD BAD BAD. Try looking at std-exten for an example of how to handle stuff when Dial exits. Script Head wrote: You can simply put then in order in your dial plan: exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED] exten = _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED] and so on. if carrier1 returns an error, * will dial out using carrier2. SciptHead On 12/2/05, *Max Clark* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I would like to have two asterisk servers in a cluster. From what I understand using a mysql database I can store all of my peer/user information in the db and share this between servers. I can then take my polycom phone and register it to both of the asterisk servers at the same time - so if one were to go offline traffic would be redirected to the second. This works in theory for the end user - but how do I provide redundancy with my upstream providers? I.e. how do I fail over my registration to an upstream sip provider? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://clarksys.com http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com http://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
[Asterisk-Users] Getting started with Asterisk and Aastra 9133i
I have a Aastra 9133i phone and would like to do a simple test to make sure everything works. I already assigned an IP address to the phone (I'm able to ping it.) I have Asterisk running (installed Asterisk and Zaptel only) but not configured. I don't have a FXS/FXO card yet but I would like to test out the phone. Ideally, I'd like to be able to setup a mailbox, record a mailbox greeting, and play it back. How do I do this? I ran make samples to install the basic config files then: I added this to the sip.conf: [aastra] type=friend host=192.168.0.99 [EMAIL PROTECTED] I also added this to the extensions.conf file: Exten = 1234,1,Wait(2) Exten = 1234,2,Record(/tmp/asterisk-recording:gsm) Exten = 1234,3,Wait(2) Exten = 1234,4,Playback (/tmp/asterisk-recording) Exten = 1234,5,wait(2) Exten = 1234,6,Hangup When I dial 1234, nothing happens. I'm not sure if that's how it supposed to work or what. ___ --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] ISDN 2e Cards
On Mon, 5 Dec 2005, Avi Miller wrote: John Daragon wrote: I'd second that. For a single ISDN2e connection the AVM Fritz card is really hard to beat/ Yeah, single is the key word there. I have 2x ISDN2 (OnRamp2 in Australia) and the AVM Fritz cards are a nightmare. Replaced the two cards with an Eicon Diva V-4BRI (so I have two extra ports if necessary) and my Asterisk box is just incredible now: Almost zero echo across the board and much lower processor utilisation. With 'echo across the board', do you mean a connection bridged between ports of the 4BRI only? If yes, there shouldn't be any echo when using line interconnect (CAPI native bridging). If this doesn't work for you, please let me know. Armin ___ --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
[Asterisk-Users] Sending data over ZAPHFC D-channel?
Is it possible to send data over the D Channel using ZAPHFC? I'd like to send data between three servers (only one is live yet, but I am thinking ahead and trying to plan...) to verify that each of their ISDN connections is live. Ie: 1 sends to 2 1 sends to 3 2 sends to 1 2 sends to 3 3 sends to 1 3 sends to 2 If this is possible, I could write an AGI script to notify on loss of ISDN link... TIA -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] ISDN 2e Cards
Armin Schindler wrote: With 'echo across the board', do you mean a connection bridged between ports of the 4BRI only? If yes, there shouldn't be any echo when using line interconnect (CAPI native bridging). If this doesn't work for you, please let me know. Heh, no -- I meant there's no (or very little) echo at all. To any connection, inbound or outbound, from the Diva card. I've called other ISDN users, analog users, rural users and it all sounds great. :) cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 2 6233 0607 Fitzroy, VIC F: +61 (0) 2 6233 0696 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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] Sipura 3000 Disconnect Singnel
Hi, I had the same problem... I've solved it by recording desconnect tone line is sending and then do frequency analysis and then you can specify custom disconnect tone on sipura 3000 configuration Procedure is described in more details on voxilla web page.. HTH, regards, Rob. - Original Message - From: Code Lover [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, December 04, 2005 10:25 AM Subject: [Asterisk-Users] Sipura 3000 Disconnect Singnel Hi all, I was testing the FXO system from sipura 3000 with asterisk PERL AGI. But when we hangup the FXO phone the channel is not disconnecting and the destination is continue ringing. even if we try to press the disconnect button for destionations after some seconds again it start to ringing. Is it problem from Sipura itself or i have to do some advance in asterisk configuration. You suggestion will be high appricated. -- Thank You, Code Lover ___ --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
Re: [Asterisk-Users] Failover Registration
Script Head wrote: You can simply put then in order in your dial plan: exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED] exten = _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED] and so on. if carrier1 returns an error, * will dial out using carrier2. SciptHead On 12/2/05, *Max Clark* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I would like to have two asterisk servers in a cluster. From what I understand using a mysql database I can store all of my peer/user information in the db and share this between servers. I can then take my polycom phone and register it to both of the asterisk servers at the same time - so if one were to go offline traffic would be redirected to the second. This works in theory for the end user - but how do I provide redundancy with my upstream providers? I.e. how do I fail over my registration to an upstream sip provider? Thanks in advance, Max Just be aware there can be lots of different reasons for calls not be processed that will not be detected and handled with the above. ___ --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
[Asterisk-Users] DISA function
Hi all, I was wondering whether the DISA function on the latest asterisk 1.2 stable release actually works better than the other prior releases. Basically the [EMAIL PROTECTED] version 2.0 BETA 4 I'm using does not recognise the DTMF tones all the time and sometime when it does, it disconnects. Cheers, Richard. ___ --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
[Asterisk-Users] UK DID 0208 £1 per month
UK, London Based DID £1 per month All number begin with 0208 0xx If you are interested please email ukdid AT cyber-telecom.net SIP based and support standard ulaw or alaw. Unlimited incoming minutes. For multi channels please email for pricing. Sam ___ --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
[Asterisk-Users] GSM Gateway / Terminal for sale
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna. Brand new unit and all of them will be tested before dispatch. Extremely easy to setup and can be used out of the box without any configuration. So should be good alternatively of phonecell or nokia pbx etc.. Units are located in UK and £60 GBP per unit excluding shipping. I have limited stock therefore please act quick to avoid disappointment Working mode: GSM 900 MHz or GSM 1800MHz double frequency Peak power: 2 W Power consume: static state 25mA, launch 600mA Sencitivity:-104dB Inner pressure :DC 12V/1.5A Condition temperature:0C~+40C Working humidity:45%-90% Atmosphere pressure:86~106Pka Circumstance noise:60 dB Wireless decibel :3.5dB or 12dB AC power:220V ac+-10%,frequency 47-54Hz;110Vac/60Hz(optional) Power port: China, USA, UK, (by customer s optional) Connection means:RJ-11 telephone line plug Antenna connection: SMA antenna tie-in, N type port(optional).TNC port(optional) For more info please email gsm AT cyber-telecom.net for more info or visit www.cyber-telecom.net to purchase right away. Sam ___ --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] New to [EMAIL PROTECTED]
You have to have a Digium TDM400, a clone X100P (eBay, about $10), or Sipura SPA-3000. A regular phone modem will not work. Another way is to get a cheap VOIP account like Broadvoice for $10 a month, then you don't need a regular phone line for the kids and you can have two calls going at the same time. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Sunday, December 04, 2005 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] New to [EMAIL PROTECTED] I'm thinking of Installing [EMAIL PROTECTED] on my PC, to contol and route calls to each of my children's computer via SoftPhone X-lite. I downloaded the ISO image, and familiar with the process to install Asterisk now. However I don't have a digium modem. Can I use any regular phone modem for incoming line on the Asterisk server? ---Dakota the Newbie ___ --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
Re: [Asterisk-Users] ISDN 2e Cards
Avi Miller wrote: John Daragon wrote: I'd second that. For a single ISDN2e connection the AVM Fritz card is really hard to beat/ Yeah, single is the key word there. I have 2x ISDN2 (OnRamp2 in Australia) and the AVM Fritz cards are a nightmare. Replaced the two cards with an Eicon Diva V-4BRI (so I have two extra ports if necessary) and my Asterisk box is just incredible now: Almost zero echo across the board and much lower processor utilisation. cYa, Avi Absolutely right. I have managed to get two of these cards running correctly in one of my machines thanks to the instructions on voip-info, but I can't say I'd be able to easily reproduce it -- I seem to remember I had to fiddle around with the drivers for ages and ages :-( Faris. ___ --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 1.2 and weird ZAP interface behaviour
You have a stray immediate=yes. Try this: grep -v ; /etc/asterisk/zapata.conf Remco Barende wrote: I just upgraded my config from * 1.0.10 to 1.2 I removed caller ID from my configs because when I try to use CallerID (new style) on my IAX provider (magrathea) but whenever I try to make a call I get a message from the provider that You are not registered to use this service. Removing the callerid stuff seems to solve this. I guess they are not ready for the new updated IAX protocol? Anyways, now to my real problem. I have a TDM11B card. Obviously one connection to the phone line, one connection to an analog phone. I just used the exact same config files as with * 1.0.10 I have this in my /etc/asterisk/zapata.conf: callerid=202 signalling=fxo_ks group=1 context=intern-all channel = 1 Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Okay, but I want to make an OUTGOING call so I don't need this phone to be in default context, do I?? When I add the default context (with s extension) to intern-all whenever I pick up the analog phone it starts ringing my default context like a bat phone. Nice but not what I wanted.. I just want it to give me a dial tone and wait for the number I want to dial. What am I overlooking here?? ___ --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
Re: [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour
You don't want immediate=yes Remco Barende wrote: On Sat, 3 Dec 2005, Begumisa Gerald M wrote: On Sat, 3 Dec 2005, Remco Barende wrote: Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Have you by chance set immediate to yes? IIRC, there's a feature that will send you to the configured context as soon as you pick up your phone (this is in zapata.conf). Might be worth checking that out. I have but only for the phone line, it is immediately after: signalling=fxs_ks immediate=yes I did some further testing, this happens only after I have done a RELOAD on the console. When I exit asterisk and start asterisk again all seems to be working as normal. Maybe it's an * bug? ___ --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
Re: [Asterisk-Users] New to [EMAIL PROTECTED]
Better yet get a Telasip account at telasip.com On 12/4/05, Kerry Garrison [EMAIL PROTECTED] wrote: You have to have a Digium TDM400, a clone X100P (eBay, about $10), or Sipura SPA-3000. A regular phone modem will not work. Another way is to get a cheap VOIP account like Broadvoice for $10 a month, then you don't need a regular phone line for the kids and you can have two calls going at the same time. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Sunday, December 04, 2005 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] New to [EMAIL PROTECTED] I'm thinking of Installing [EMAIL PROTECTED] on my PC, to contol and route calls to each of my children's computer via SoftPhone X-lite. I downloaded the ISO image, and familiar with the process to install Asterisk now. However I don't have a digium modem. Can I use any regular phone modem for incoming line on the Asterisk server? ---Dakota the Newbie ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --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] Broadband VoIP Startup with Asterisk
[EMAIL PROTECTED] wrote: We are currently an ISP offering broadband and wireless internet connections. We are planning to start offering VoIP services to our current customers. We have decided to use Asterisk as our PBX software. If I may, I have a multi-part question. 1) If we do use Asterisk, what would be a good billing system to use to keep track of thousands of customers minutes? 2) Are there are any good back end software available for Asterisk in which our customers could log into and view their usage, etc? 3) We are currently testing the Asterisk system using VoIPJet over an IAX connection to our Asterisk system. Would their be any benefits or disadvantages to putting in a PRI line instead? 4) What is some good company names to purchase DID's and VoIP termination from? I have been looking at VoIPJet and Teliax. I also am a member of DIDX. DIDX does not seem to have any numbers in the area where I will be marketing first. I have read some solutions to all my answers above on the Wiki, however, I would like some comments from people who have actually gone through his process. As we all know, it is much easier to put in a good system from the beginning, then have to switch everything over once its running. Thanks for everyone help with the above questions. -jglucky ___ --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 As as WISP you should be familiar with http://www.part-15.org. If not, they are a Wireless Internet Service Providers Orginization. They have a reseller program for VOIP and offer different monthly commision rates for members and non members of the orginization. You can get more information from this address. http://www.part-15.org/voip/reseller.htm This could be an alternative to rolling your own solution. Just a thought. Dan ___ --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
[Asterisk-Users] diax not working properly
Hi! I have been using Asterisk-1.0.3 for quite some time now.My main aim nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The problem is that sometimes the phone doesn`t register and at others it gets out of the registration(after being registere for some time).Can anyonetell me what can be the problem ,what other iax phones are available ? Thanx and Regards, Amna ___ --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
[Asterisk-Users] Connecting 2 Asterisk using SIP
I have 2 Asterisk servers running 1.2.0. One of them is a PSTN gateway. Currently they are connected using IAX2. I wanted to play with SIP. I setup a sip entry (type=friend) in the PSTN gateway box and a sip entry (type=user) in the second box in order to send calls using SIP to the second box. This works fine. However, when I setup the second box as type=friend in order for it to be able to send calls back to the gateway box, then calls no longer work from gateway box to the second box. The reported error is: Dec 5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite: Failed to authenticate on INVITE to '2125551212 sip: [EMAIL PROTECTED];tag=as0698b1b9' In the gateway box, my sip.conf looks like this: [general] allowguest=yes autocreatepeer=no [secondbox] type=friend host=10.0.0.2 secret=mysecret In the second box, my sip.conf looks like this: [general] allowguest=yes autocreatepeer=no [secondbox] type=user host=10.0.0.1 secret=mysecret Any ideas on how to correctly set this up? Thanks, Waldo ___ --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] ZapHFC cards not maintaining sync?!
Already HAVE Florz patch installed! :-( What version of * and BRIstuff are you using? Strange, sounds like the florz patch has not been effectively applied or it's broken. I'm using an old version of bristuff : Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n built by [EMAIL PROTECTED] on a x86_64 running Linux I had an issue compiling it for x86_64 though, but that's a different question. I assumed as much when I saw your last name... :-) Whereabouts in NL? I'm in Zoetermeer (ZH)... Amsterdam, but the ISDN setup I installed near Leiden (ZH) :) 1) Every 10 seconds () the D channel gets torn down, which That's too slow, it should happen about every 1-2 seconds or so. The d channel going down and up again is normal behaviour. I know it is. Used to work for a Networking Competence Centre, and we had the same kind of issues with 3Com Netbuilders. The first call attempt after the D Channel was torn down always failed... The only solution was to get KPN to turn on the D Channel permanently... Strange, I never had that problem before. When the * box gets up I can immediately make calls. Also the standard KPN A/B equipment doesn't have this problem, sounds like it's more 3Com related. One problem I have found with bristuff (and no solution yet), if you disconnect the ISDN line from the * box (or the ISDN line is out of order for a short while), bristuff will not re-establish the connection. It is then required to unload all the modules and re-load them or even worse, reboot the box. I guess that is a specific bristuff problem. All calls to the ISDN line fail and it's not possible to make any calls. Even after several hours bristuff doesn't setup the line connection. 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... I noticed that the 'deactivated' issue doesn't happen for a while after a call has been placed. I am now testing placing a call every minute, with a 100 ms timeout using the manager api. This means it never actually gets a chance to get through, or be picked up, but it does cause activity on the D channel. This has been running for half an hour now, and I haven't seen the channel go down for extended periods since. I'm not sure whether the KPN will like it, but it's an interesting test to run! G Good luck with our Royal Dutch KPN, but I would try florz first :) Tell me about it! Like I said above, we had *extensive* experience with them over the D Channel issue! Weird, I checked with KPJ before and he mentioned it is normal behaviour for ISDN. My console is filled with messages like this : == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down and it doesn't cause me any issues. It would be nice to 'hide' these messages when not in very verbose mode to avoid cluttering up the console. The messages indeed do appear about every 10 seconds or so. ___ --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
[Asterisk-Users] Sipura 3000 Call waiting on the PSTN line
I have configured my SPA-3000 to direct all calls received over PSTN interface to Asterisk. That is set up via a dial plan with S0 and works fine. CID information is passed along as well. But I find it impossible to setup call waiting on the PSTN line. I do subscribe to this service from my PSTN provider (SBC). Once a PSTN call comes in, is forwarded to asterisk and then to a phone at line 1, Sipura seems oblivious to any other calls coming in on PSTN. When I dial my PSTN number with a second call I don't see any activity on Sipura. I have set up remote logging and the level is set to 3. But I still get no activity in the log for the second call. Even the first call logs very little, but it logs something. Has anybody been able to get call waiting on the PSTN line? Thanks, Chip ___ --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] Sipura 3000 Call waiting on the PSTN line
Has anybody been able to get call waiting on the PSTN line? As far as I recall, you will only hear a tone in the audio stream when a second call comes in. The Sipura does not detect or handle it, but if you flash the line on the FXS interface after hearing the tone, the Sipura will forward the flash to the FXO interface and hence switch to the second call. I am positive this works when the call is picked up on the local FXS port but I am not sure if it also works when the call is picked up by a remote device. This is how I had set it up: PSTN - FXO on Sipura 3K - Asterisk (remote via DSL) - FXS on Sipura 3K - Phone The call would be re-invited in this case so no RTP traffic goes via DSL, only SIP traffic. Switching to second call with flash works in this scenario. Additionally I also allowed the call to be received by a remote device (RTP via DSL) but I am not sure if you can then use Call Waiting (never tried it). I don't think I'm expressing myself clearly here; if not, please ask. Or correct me if I'm wrong. Luki ___ --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] Connecting 2 Asterisk using SIP
Any ideas on how to correctly set this up? Try adding authuser= and/or username= to the configuration. Do a SIP DEBUG and see what peer asterisk looks for when trying to authenticate the INVITE. It probably can't find the right peer; authuser on the initiating end should help in this case. --Luki ___ --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
[Asterisk-Users] Re: sound problem in X-Lite phone with asterisk server
Hi all I am a newbie to the asterisk. I just installed asterisk server and two X-Lite softphones. I allready configured sip.conf and extension.conf. Now when i call from one softphone to other , sip signaling is going perfect. Both phone are in ringing mode. But i can't able to hear ring. When i pickup call, there is not any sound at all. The asterisk server give following output during call: Dec 5 12:49:57 NOTICE[1931]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Dec 5 12:49:57 WARNING[1931]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' Dec 5 12:49:57 WARNING[1931]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player Can any one pls tell me where i am going wrong. Thanks Vipul ___ --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
[Asterisk-Users] [Amportal-users] AMP queues, AddQueueMember and 'Wrapup Time'
Hi all, I posted about this briefly, but haven't gotten a response as yet. I've checked further, and it seems that the 'Wrapup Time' option just isn't having any effect. We have agents logging into the queue using [queueno]* and [queueno]** to log out. The calls are all processed fine, but it just seems to hit them straight away again, not obeying any of the settings in wrapup. Has anyone else seen this? or does anyone have any suggestions as to where to start looking ? Thanks -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --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
[Asterisk-Users] voicexml vendors
Any suggestions/comments on companies that provide hosted voicexml solutions that work with SIP? Seems like a new enough market that the pricing is pretty high and the number of vendors that will work with smaller volumes is low. So far Voxeo is at the top of my list. Chris ___ --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
[Asterisk-Users] Error when compiling asterisk
Hi, I am getting this error when compiling asterisk `ls *.c`: unrecognized optionh -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ...GNU long options: --debug --dump-po-strings --dump-strings --help --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexpShell options: -irsD or -c command (invocation only) -abefhkmnptuvxBCHP or -o optionmake: *** [.depend] Error 2 Any ideas of what the problem might be. Thank you Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez le ici ! ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Biz mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-biz Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez le ici ! ___ --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
