Re: [Asterisk-Users] Meetme Timing Interface
Douglas Garstang wrote: > I have ztdummy installed: > > Module Size Used by > ztdummy 3464 0 > zaptel218756 1 ztdummy > crc_ccitt 2176 1 zaptel > ohci_hcd 16388 0 > floppy 49028 0 > pcspkr 2180 0 > piix8580 0 [permanent] > ehci_hcd 24456 0 > uhci_hcd 26256 0 > rtc10164 1 ztdummy > usbcore84740 4 ohci_hcd,ehci_hcd,uhci_hcd > > However, when I enter a meetme conference, I get this: > > -- Playing 'conf-getconfno' (language 'en') > Mar 3 15:27:26 WARNING[23657]: channel.c:2535 ast_request: No channel type > registered for 'zap' > Mar 3 15:27:26 WARNING[23657]: app_meetme.c:461 build_conf: Unable to open > pseudo channel - trying device > -- Created MeetMe conference 1023 for conference '123' > > Uhm WHY? If I didn't have ztdummy installed, Asterisk would complain that > my conference number is not valid, and I would see errors on Asterisk startup > about not being able to find a timing interface. These things are not > happening. However, it is spitting out that error message on the console. Why? You need to compile asterisk after compiling zaptel. Otherwise chan_zap.so won't get created. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Changing caller id on transfer
On 03/04/06 23:17 Cosmin Prund said the following: My dial plan is as simple as it gets: exten => 101,1,Dial(sip/sip101,180,Ttr) But I'm doing blind transfers and you're doing attended transfers. oh right, i had misadverntly thought you were doing attended xfers as well. with blind xfers, we do get the behaviour you've noticed. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ --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 1.2.5 Released
On 03/04/06 23:54 The Asterisk Development Team said the following: However, there is also a patch against the previous release as an option for a smaller download, asterisk-1.2.5-patch.gz. well done, this makes it a lot easier on the downloads for those closely tracking the releases. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ --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] Child PID's
On 03/04/06 16:30 Paul Hewlett said the following: On 2.4 kernels you would be using the LinuxThreads implementation of POSIX threads. This emulated the POSIX threading model with some limitations - to continue with this thread (pun intended !) and for freebsd users, the default asterisk as downloaded builds nicely against libpthreads, but you could try the alternative libthr.so.1 by switching them around /post build/. just copy libpthread.so.1 to another location and copy libthr.so.1 as libpthread.so.1. this allows you to test threading speed in both the libraries. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ --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] Program Buttons on Cisco 79xx Phones
On Sat, 4 Mar 2006, Greg Oliver wrote: On Sat, 2006-03-04 at 10:34 +, Ron Wellsted wrote: Unfortunately you have to make a choice: SIP firmware - Easy to implement on *, but poor XML support SCCP firmware - poor/non-trivial asterisk support, great XML support. The newest SIP firmware (beta versions) allows the exact XML functionality as the SCCP versions. Since Cisco CME and CCM are both migrating to SIP (CME already has) in their next versions, the loads provide all the bells and whistles. We're still waiting for a SIP-enabled 7970... The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step backwards to me. -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
RE: [Asterisk-Users] Program Buttons on Cisco 79xx Phones
Sounds great..thanks... -Original Message- From: Greg Oliver [mailto:[EMAIL PROTECTED] Sent: Saturday, March 04, 2006 8:25 PM To: [EMAIL PROTECTED]; Asterisk User List Subject: Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones On Sat, 2006-03-04 at 10:34 +, Ron Wellsted wrote: > Unfortunately you have to make a choice: > SIP firmware - Easy to implement on *, but poor XML support > SCCP firmware - poor/non-trivial asterisk support, great XML support. The newest SIP firmware (beta versions) allows the exact XML functionality as the SCCP versions. Since Cisco CME and CCM are both migrating to SIP (CME already has) in their next versions, the loads provide all the bells and whistles. We have the development loads and have every Cisco model currently running in SIP. Currenltly, I cannot register to * with them since there is no auth in them yet. For example - on a 40/60, it only allows a single global logon (not per line like current SIP firmware) and that does not even work - never tries to register with asterisk at all and I only had about an hour at work to spend on it. I can say that the numbering scheme for SIP loads (7.5, etc) has remained intact. So the probable reason there has not been a 7.6 release even with all of the 7.5 bugs is because the 8.0 version is in beta/development right now. They are gonna be awesome phones once they have all the SCCP capabilities for SIP if the decide to merge the codebase into CCM/3rd-party compatibility. -Greg ___ --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] Auto dial feature
> a message and it notifies the on call techs. My question is regarding > externnotify for the voice mail.conf. If I enabled that and set up a > call file, will it do it for every voice mail box I have on the system? > Is there a way I can limit it to just the one voice mail box on the > system? If not, what would be the best way to send out the voice mail from the wiki : The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox. These arguments are passed to the program that you set in the externnotify variable. So, you could check in your script what extension the voicemail was left on (second argument) and only if it is the mailbox you want to be notified of, you create that call file. > message that was recorded to our on call techs. I need it to attempt 3 > times in two minute intervals. Any suggestions is greatly appreciated. In a call file you can specify how many retry hth ___ --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] really need help with outgoing calls..PSTN errors
You should also know that this ONLY works with DTMF on analog lines. If
one happens to have to use pulse dial on a POTS line, there is no way to
delay dialing, and Asterisk STILL will not wait for dialtone, since no
one who is able to fix it seems interested.
John Novack
sdgesa gaeharth wrote:
Thanks. I will try. Is there any documentation that describes this
fix? I cant find it anywhere in any docs.
*/Joseph Tanner <[EMAIL PROTECTED]>/* wrote:
Like this:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})
Joseph Tanner
On 3/4/06, sdgesa gaeharth wrote:
>
> You mean like this
>
> exten => ww_9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
>
> thanks
>
>
> Joseph Tanner wrote:
>
> Put a w or 2 (ww) in front of your number being dialed, it
should work
> then. If not, try more w's.
>
> Joseph Tanner
>
> On 3/3/06, sdgesa gaeharth wrote:
> >
> > I cant seem to get outgoing calls to be placed properly .. No
matter what
> I
> > try I get an error from the PSTN company saying that the "call
can not be
> > completed as dialed" or "you need to dial a one..." The
asterisk debugging
> > seems to show the correct number being dialed out of the zap
interface...
> > the "9" is being stripped and there is a "1" where it is
supposed to be. I
> > am thinking it is a problem between the zap interface and the
PSTN.
> >
> > thanks
> >
> > extensions.conf
> > [general]
> > static=yes
> > writeprotect=no
> > autofallthrough=yes
> > clearglobalvars=no
> > priorityjumping=no
> > [globals]
> > ATTENDANT=1001
> > OUTBOUNDTRUNK=ZAP/g1
> > [extentions]
> > exten => _10XX,1,Ringing
> > exten => _10XX,2,Dial(SIP/${EXTEN},20)
> > exten => _10XX,3,Answer
> > exten => _10XX,4,VoiceMail([EMAIL PROTECTED])
> > exten => _10XX,5,Hangup
> > [voicemail]
> > exten => _910XX,1,Wait(1)
> > exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
> > [local]
> > include => extentions
> > include => voicemail
> > [incoming]
> > exten => s,1,Answer
> > exten => s,n,Wait(2)
> > exten => s,n,Set(TIMEOUT(response)=15)
> > exten => s,n,Background(company-intro)
> > exten => s,n,WaitExten()
> > exten => s,n,Playback(vm-goodbye)
> > exten => s,n,Hangup()
> > exten => 0,1,Dial(SIP/${ATTENDANT},20)
> > exten => 1,1,Directory(voicemail,extentions,f)
> > exten => 2,1,Directory(voicemail,extentions)
> > exten => 1234,1,Playback(abandon-all-hope)
> > include => extentions
> > exten => i,1,Playback(vm-goodbye)
> > exten => i,2,Hangup()
> > exten => t,1,Playback(vm-goodbye)
> > exten => t,2,Hangup()
> > [outbound]
> > ignorepat => 9
> > exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _9XX,2,Congestion()
> > exten => _9XX,102,Congestion()
> > exten =>
> _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91800NXX,2,Congestion()
> > exten => _91800NXX,102,Congestion()
> > exten =>
> _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91888NXX,2,Congestion()
> > exten => _91888NXX,102,Congestion()
> > exten =>
> _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91877NXX,2,Congestion()
> > exten => _91877NXX,102,Congestion()
> > exten =>
> _91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91866NXX,2,Congestion()
> > exten => _91866NXX,102,Congestion()
> > exten => _91900NXX,1,Congestion()
> > exten => _91976NXX,1,Congestion()
> > exten =>
> > _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91[1234567]XXNXX,2,Congestion()
> > exten => _91[1234567]XXNXX,102,Congestion()
> > exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
> > exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411)
> > exten => 0,1,Dial(${OUTBOUNDTRUNK}/0)
> >
> > [local-access]
> > include => local
> > include => outbound
> >
> > zapata.conf:
> > [channels]
> > group => 1
> > language=en
> > context=incoming
> > signalling=fxs_ks
> > switchtype=national
> > usecallerid=yes
> > hidecallerid=no
> > callwaiting=yes
> > callerid => "Dulles Micro, LLC" <703 450 5000>
> > usecallingpres=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > canpark=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > rxgain=0.0
> > txgain=0.0
> > channel => 1
> >
> > zaptel.conf:
> > fxsks=1,2,3,4
> > loadzone = us
> > defaultzone=us
> >
> >
> >
> >
> >
> >
> > Brings words and
Re: [Asterisk-Users] really need help with outgoing calls..PSTN errors
More and more areas of the US require 10 digit local dialing, and 11
digit toll dialing.
Unfortunately, that isn't universally true. Some states have decreed
that 11 digits will be dialed for local and toll, other locales have 7
digit dialing across state lines, and at least one location, probably
more, require 7 digit dialing to another state for one NPA, and 11 digit
for it's overlay.
Mobile providers in the US usually require only 10 digits, and fill in
the "1" within the phone.
The only rule is there are no rules. The result of local/state rule.
John Novack
sdgesa gaeharth wrote:
In our area code(703), and I am not sure if it is like this in other
places, we are required to dial the area code even if we dial local
numbers . That is what these lines are for:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _9XX,2,Congestion()
exten => _9XX,102,Congestion()
Any other options?
*/Mark Hulber <[EMAIL PROTECTED]>/* wrote:
Have you tried dialing an 800 number? Does that work? This extension:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
seems to be missing one X since it's only 10 digits long. Your PSTN
probably requires a 1 to be dialed also. On the other hand,
exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
you should probably be matching this extension instead although you
won't be able to match anywhere that has an area code that starts
with
an 8 or 9. (905, 916, 914 as a few examples).
MARK.
sdgesa gaeharth wrote:
> I cant seem to get outgoing calls to be placed properly .. No
matter
> what I try I get an error from the PSTN company saying that the
"call
> can not be completed as dialed" or "you need to dial a one..." The
> asterisk debugging seems to show the correct number being dialed
out
> of the zap interface... the "9" is being stripped and there is a
"1"
> where it is supposed to be. I am thinking it is a problem
between the
> zap interface and the PSTN.
>
> thanks
>
> extensions.conf
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
> [globals]
> ATTENDANT=1001
> OUTBOUNDTRUNK=ZAP/g1
> [extentions]
> exten => _10XX,1,Ringing
> exten => _10XX,2,Dial(SIP/${EXTEN},20)
> exten => _10XX,3,Answer
> exten => _10XX,4,VoiceMail([EMAIL PROTECTED]
> )
> exten => _10XX,5,Hangup
> [voicemail]
> exten => _910XX,1,Wait(1)
> exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
> [local]
> include => extentions
> include => voicemail
> [incoming]
> exten => s,1,Answer
> exten => s,n,Wait(2)
> exten => s,n,Set(TIMEOUT(response)=15)
> exten => s,n,Background(company-intro)
> exten => s,n,WaitExten()
> exten => s,n,Playback(vm-goodbye)
> exten => s,n,Hangup()
> exten => 0,1,Dial(SIP/${ATTENDANT},20)
> exten => 1,1,Directory(voicemail,extentions,f)
> exten => 2,1,Directory(voicemail,extentions)
> exten => 1234,1,Playback(abandon-all-hope)
> include => extentions
> exten => i,1,Playback(vm-goodbye)
> exten => i,2,Hangup()
> exten => t,1,Playback(vm-goodbye)
> exten => t,2,Hangup()
> [outbound]
> ignorepat => 9
> exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _9XX,2,Congestion()
> exten => _9XX,102,Congestion()
> exten => _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _91800NXX,2,Congestion()
> exten => _91800NXX,102,Congestion()
> exten => _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _91888NXX,2,Congestion()
> exten => _91888NXX,102,Congestion()
> exten => _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _91877NXX,2,Congestion()
> exten => _91877NXX,102,Congestion()
> exten => _91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _91866NXX,2,Congestion()
> exten => _91866NXX,102,Congestion()
> exten => _91900NXX,1,Congestion()
> exten => _91976NXX,1,Congestion()
> exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _91[1234567]XXNXX,2,Congestion()
> exten => _91[1234567]XXNXX,102,Congestion()
> exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
> exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411)
> exten => 0,1,Dial(${OUTBOUNDTRUNK}/0)
>
> [local-access]
> include => local
> include => outbound
>
> zapata.conf:
> [channels]
> group => 1
> language=en
> context=incoming
> signalling=fxs_ks
> switchtype=national
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callerid => "Dulles Micro, LLC" <703 450 5000>
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> tra
Re: [Asterisk-Users] *** Yet another boring weekend? Test new Asterisk features in development!
Testing Asterisk doenst make for a boring day at all :) On 3/4/06, Olle E Johansson <[EMAIL PROTECTED]> wrote: > In Sweden, where I live, it's snowing like crazy. The Stockholm area > is covered in white stuff > and there's really no reason to leave the computer and get out > anywhere. More white stuff > is coming down all the time. Boring. I am sure your weekend is no > better - rain, snow or > just another boring sunny day. > > Let's find something cool to do during this weekend! > > Join the cool crowd that tests the test branch during evenings and > weekends. The dudes and dudettes that > proudly contributes by reporting everything from simple spelling > errors to crashes and strange noices from > their Asterisk boxes. The people who knows what is going on in the > Asterisk development circles - the > Asterisk Test Team! > > I've updated the test branch to the latest version of my SIP > peermatch code. This is quite a large code > change, but not as large a functional change. However, it changes > some basic functionality: > > * The sip_user structure is gone > * Incoming calls are matched first by user from: name, then peer > From: name, then IP. > * Friends are now *one* in-memory object. > > In most cases, this means you can change type=friend to type=peer for > local phones on the > same LAN. This will also improve SIP subscriptions (blinking lights) > and call limits, since for > both friends and peers, we now have *one* object in memory that > handles the limit for both incoming > and outgoing calls. > > During the week, I've also added a few other patches by other > contributors. > > Read the README.test-this-branch here: > http://svn.digium.com/view/asterisk/team/oej/test-this-branch/ > README.test-this-branch?view=markup > > ** PLEASE help the community, please test this branch. > > Check it out like this > > svn checkout http://svn.digium.com/svn/asterisk/team/oej/test-this- > branch test-trunk > > Then cd into test-trunk and run "make" then "make install" > > Report any bugs in the proper open bug in the bug tracker. If you > like new functions, add a comment that this works for you. Provide > feedback, make our work easier. > > Run "svn update" from time to time to get the latest version. Any > changes from trunk will be merged into this code. Read the > README.test-this-branch file to get more information. > > Thank you for your help! > > /Olle > > PS. Obviously, this is test code, not recommended to be closer than 2 > miles (20 kilometers) from your production servers. > ___ > --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] Preferred editor(s) dialplan coding?
vi here On 3/4/06, JP Carballo <[EMAIL PROTECTED]> wrote: > > Bill Gibbs wrote: > > >Vim for everything > > > >-Original Message- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL PROTECTED] On Behalf Of Pete > >Barnwell > >Sent: Friday, March 03, 2006 7:39 PM > >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] Preferred editor(s) dialplan coding? > > > >Emacs... > > > >On Sat, 2006-03-04 at 01:35 +0100, adibar wrote: > > > > >Vim forever ;-) > > > > > >On Fri, Mar 03, 2006 at 03:06:02PM -0500, S McGowan wrote: > > > > > emacs for me :) > > -- > JP Carballo > > http://www.netfone2x.com > Bringing the world closer. > > It might look like I'm doing nothing, but at the cellular level, I'm really > quite busy. > > ___ > --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] incoming calls dropout on PRI over TE110p
Funnily - I have set up 2 or 3 pri's over the last few weeks on 1.2x and haven't had any issues. (and one of those is a high load situation - passthru at an outbound call centre) PaulH Melbourne - Original Message - From: "James Sturges" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Sunday, March 05, 2006 7:52 AM Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p > I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of > the site. It is sending CRC errors )to Telsta, drops all calls once a day > for 1 second, calls getting stuck, quite unpleasant! > > I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri. > > James > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Paul C > Sent: Wednesday, 1 March 2006 4:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p > > > Paul C wrote: > >> I am running Asterisk 1.0.9 and have been running all my calls through a > >> VSP over a IAX2 trunk however we have recently purchased and connected a > >> TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make > > >> outgoing calls via it fine, however incoming calls are dropped after a > >> few seconds ( or as soon as a command like Playback, or the call is > >> picked up if forwarded to a SIP extensions ). > > >> SNIP << > > > > > overlapdial should usually be no in my experience. > > > Okay I've turned that to no with no change. I've just got off the phone to > Optus and apparently they had a client in melbourne last week and they fixed > > the problem by turning crc checking off at the optus end. I don't suppose > that was anybody on 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 > ___ --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] really need help with outgoing calls..PSTN errors
In our area code(703), and I am not sure if it is like this in other places, we are required to dial the area code even if we dial local numbers . That is what these lines are for: exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})exten => _9XX,2,Congestion()exten => _9XX,102,Congestion() Any other options? Mark Hulber <[EMAIL PROTECTED]> wrote: Have you tried dialing an 800 number? Does that work? This extension:exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})seems to be missing one X since it's only 10 digits long. Your PSTN probably requires a 1 to be dialed also. On the other hand,exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})you
should probably be matching this extension instead although you won't be able to match anywhere that has an area code that starts with an 8 or 9. (905, 916, 914 as a few examples).MARK.sdgesa gaeharth wrote:> I cant seem to get outgoing calls to be placed properly .. No matter > what I try I get an error from the PSTN company saying that the "call > can not be completed as dialed" or "you need to dial a one..." The > asterisk debugging seems to show the correct number being dialed out > of the zap interface... the "9" is being stripped and there is a "1" > where it is supposed to be. I am thinking it is a problem between the > zap interface and the PSTN.> > thanks> > extensions.conf> [general]> static=yes> writeprotect=no> autofallthrough=yes> clearglobalvars=no> priorityjumping=no> [globals]> ATTENDANT=1001>
OUTBOUNDTRUNK=ZAP/g1> [extentions]> exten => _10XX,1,Ringing> exten => _10XX,2,Dial(SIP/${EXTEN},20)> exten => _10XX,3,Answer> exten => _10XX,4,VoiceMail([EMAIL PROTECTED] > )> exten => _10XX,5,Hangup> [voicemail]> exten => _910XX,1,Wait(1)> exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])> [local]> include => extentions> include => voicemail> [incoming]> exten => s,1,Answer> exten => s,n,Wait(2)> exten => s,n,Set(TIMEOUT(response)=15)> exten => s,n,Background(company-intro)> exten => s,n,WaitExten()> exten => s,n,Playback(vm-goodbye)> exten => s,n,Hangup()> exten => 0,1,Dial(SIP/${ATTENDANT},20)> exten => 1,1,Directory(voicemail,extentions,f)> exten => 2,1,Directory(voicemail,extentions)> exten =>
1234,1,Playback(abandon-all-hope)> include => extentions> exten => i,1,Playback(vm-goodbye)> exten => i,2,Hangup()> exten => t,1,Playback(vm-goodbye)> exten => t,2,Hangup()> [outbound]> ignorepat => 9> exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> exten => _9XX,2,Congestion()> exten => _9XX,102,Congestion()> exten => _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> exten => _91800NXX,2,Congestion()> exten => _91800NXX,102,Congestion()> exten => _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> exten => _91888NXX,2,Congestion()> exten => _91888NXX,102,Congestion()> exten => _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> exten => _91877NXX,2,Congestion()> exten => _91877NXX,102,Congestion()> exten =>
_91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> exten => _91866NXX,2,Congestion()> exten => _91866NXX,102,Congestion()> exten => _91900NXX,1,Congestion()> exten => _91976NXX,1,Congestion()> exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> exten => _91[1234567]XXNXX,2,Congestion()> exten => _91[1234567]XXNXX,102,Congestion()> exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)> exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411)> exten => 0,1,Dial(${OUTBOUNDTRUNK}/0)>> [local-access]> include => local> include => outbound> > zapata.conf:> [channels]> group => 1> language=en> context=incoming> signalling=fxs_ks> switchtype=national> usecallerid=yes> hidecallerid=no> callwaiting=yes> callerid => "Dulles Micro, LLC" <703 450
5000>> usecallingpres=yes> callwaitingcallerid=yes> threewaycalling=yes> transfer=yes> canpark=yes> cancallforward=yes> callreturn=yes> echocancel=yes> echocancelwhenbridged=yes> rxgain=0.0> txgain=0.0> channel => 1> > zaptel.conf:> fxsks=1,2,3,4> loadzone = us> defaultzone=us> > > >> > Brings words and photos together (easily) with> PhotoMail > > - it's free and works with Yahoo! Mail.> >> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> T
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Re: [Asterisk-Users] really need help with outgoing calls..PSTN errors
Thanks. I will try. Is there any documentation that describes this fix? I cant find it anywhere in any docs.Joseph Tanner <[EMAIL PROTECTED]> wrote: Like this:exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})Joseph TannerOn 3/4/06, sdgesa gaeharth <[EMAIL PROTECTED]>wrote:>> You mean like this>> exten => ww_9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})>> thanks>>> Joseph Tanner <[EMAIL PROTECTED]>wrote:>> Put a w or 2 (ww) in front of your number being dialed, it should work> then. If not, try more w's.>> Joseph Tanner>> On 3/3/06, sdgesa gaeharth wrote:> >> > I cant seem to get outgoing calls to be placed properly .. No matter
what> I> > try I get an error from the PSTN company saying that the "call can not be> > completed as dialed" or "you need to dial a one..." The asterisk debugging> > seems to show the correct number being dialed out of the zap interface...> > the "9" is being stripped and there is a "1" where it is supposed to be. I> > am thinking it is a problem between the zap interface and the PSTN.> >> > thanks> >> > extensions.conf> > [general]> > static=yes> > writeprotect=no> > autofallthrough=yes> > clearglobalvars=no> > priorityjumping=no> > [globals]> > ATTENDANT=1001> > OUTBOUNDTRUNK=ZAP/g1> > [extentions]> > exten => _10XX,1,Ringing> > exten => _10XX,2,Dial(SIP/${EXTEN},20)> > exten => _10XX,3,Answer> > exten =>
_10XX,4,VoiceMail([EMAIL PROTECTED])> > exten => _10XX,5,Hangup> > [voicemail]> > exten => _910XX,1,Wait(1)> > exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])> > [local]> > include => extentions> > include => voicemail> > [incoming]> > exten => s,1,Answer> > exten => s,n,Wait(2)> > exten => s,n,Set(TIMEOUT(response)=15)> > exten => s,n,Background(company-intro)> > exten => s,n,WaitExten()> > exten => s,n,Playback(vm-goodbye)> > exten => s,n,Hangup()> > exten => 0,1,Dial(SIP/${ATTENDANT},20)> > exten => 1,1,Directory(voicemail,extentions,f)> > exten => 2,1,Directory(voicemail,extentions)> > exten => 1234,1,Playback(abandon-all-hope)> > include => extentions> > exten => i,1,Playback(vm-goodbye)> > ex
ten
=> i,2,Hangup()> > exten => t,1,Playback(vm-goodbye)> > exten => t,2,Hangup()> > [outbound]> > ignorepat => 9> > exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _9XX,2,Congestion()> > exten => _9XX,102,Congestion()> > exten =>> _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91800NXX,2,Congestion()> > exten => _91800NXX,102,Congestion()> > exten =>> _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91888NXX,2,Congestion()> > exten => _91888NXX,102,Congestion()> > exten =>> _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91877NXX,2,Congestion()> > exten => _91877NXX,102,Congestion()> > exten =>>
_91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91866NXX,2,Congestion()> > exten => _91866NXX,102,Congestion()> > exten => _91900NXX,1,Congestion()> > exten => _91976NXX,1,Congestion()> > exten =>> > _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91[1234567]XXNXX,2,Congestion()> > exten => _91[1234567]XXNXX,102,Congestion()> > exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)> > exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411)> > exten => 0,1,Dial(${OUTBOUNDTRUNK}/0)> >> > [local-access]> > include => local> > include => outbound> >> > zapata.conf:> > [channels]> > group => 1> > language=en> > context=incoming> > signalling=fxs_ks> > switchtype=national>
>
usecallerid=yes> > hidecallerid=no> > callwaiting=yes> > callerid => "Dulles Micro, LLC" <703 450 5000>> > usecallingpres=yes> > callwaitingcallerid=yes> > threewaycalling=yes> > transfer=yes> > canpark=yes> > cancallforward=yes> > callreturn=yes> > echocancel=yes> > echocancelwhenbridged=yes> > rxgain=0.0> > txgain=0.0> > channel => 1> >> > zaptel.conf:> > fxsks=1,2,3,4> > loadzone = us> > defaultzone=us> >> >> >> >> >> > > > Brings words and photos together (easily) with> > PhotoMail - it's free and works with Yahoo! Mail.> >> >> > ___> > --Bandwidth and Colocation provide
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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> > Yahoo! Mail> Use Photomail to share photos without annoying attachments.>>> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>>
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[Asterisk-Users] Auto dial feature
Hey everyone, We have a special mail box for certain customers when we are out of the office. Basically they enter a pin number and if it is valid they leave a message and it notifies the on call techs. My question is regarding externnotify for the voice mail.conf. If I enabled that and set up a call file, will it do it for every voice mail box I have on the system? Is there a way I can limit it to just the one voice mail box on the system? If not, what would be the best way to send out the voice mail message that was recorded to our on call techs. I need it to attempt 3 times in two minute intervals. Any suggestions is greatly appreciated. Thanks, Kevin ___ --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] Program Buttons on Cisco 79xx Phones
On Sat, 2006-03-04 at 10:34 +, Ron Wellsted wrote: > Unfortunately you have to make a choice: > SIP firmware - Easy to implement on *, but poor XML support > SCCP firmware - poor/non-trivial asterisk support, great XML support. The newest SIP firmware (beta versions) allows the exact XML functionality as the SCCP versions. Since Cisco CME and CCM are both migrating to SIP (CME already has) in their next versions, the loads provide all the bells and whistles. We have the development loads and have every Cisco model currently running in SIP. Currenltly, I cannot register to * with them since there is no auth in them yet. For example - on a 40/60, it only allows a single global logon (not per line like current SIP firmware) and that does not even work - never tries to register with asterisk at all and I only had about an hour at work to spend on it. I can say that the numbering scheme for SIP loads (7.5, etc) has remained intact. So the probable reason there has not been a 7.6 release even with all of the 7.5 bugs is because the 8.0 version is in beta/development right now. They are gonna be awesome phones once they have all the SCCP capabilities for SIP if the decide to merge the codebase into CCM/3rd-party compatibility. -Greg ___ --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] seg fault when skinny phone answers
Downgrade to 1.0.10. I was unable to get the 12sp+ to work reliably in 1.2.0-1.2.4 and had the same problem.On 2/20/06, btb <
[EMAIL PROTECTED]> wrote:hello-i'm having trouble completing a connection between an older skinny
phone (12sp+) and a soft sip phone (x-lite).the skinny phone appears to successfully register: -- Starting Skinny session from 192.168.1.50Device SEP00D0BA03AB66 is attempting to register
-- Device 'office' successfuly registeredRequesting capabilitiesVersion RequestReceived CapabilitiesResButtontemplate requestedSending 12SP template to [EMAIL PROTECTED] (12SP)Received Time/Date Request
when a place a call from x-lite, the 12sp+ rings, and asterisk says: -- Executing Dial("SIP/ion-226a", "Skinny/[EMAIL PROTECTED]|20|tr") innew stackFound device: office -- skinny_request(
[EMAIL PROTECTED]) -- Skinny cw: 0, dnd: 0, so: 0, sno: 0skinny_new: tmp->nativeformats=4 fmt=4 -- skinny_call(Skinny/[EMAIL PROTECTED])Trying to send: 2rämó@'Displaying message 2rämó@'Displaying Prompt Status 'Ring-In'
-- Called [EMAIL PROTECTED] -- Skinny/[EMAIL PROTECTED] is ringingas soon as i answer the call (or hangup from x-lite, or wait for thetimeout period), aterisk says: -- Skinny/[EMAIL PROTECTED] answered SIP/ion-226a
Segmentation fault (core dumped)in addition, i can't make a call from the 12sp. when i dial x-litefrom the 12sp, asterisk says:Attempting to Clear display on Skinny [EMAIL PROTECTED]skinny_new: tmp->nativeformats=4 fmt=4
-- Starting simple switch on '[EMAIL PROTECTED]'Collected digit: [8] -- Asked to indicate 'Stop tone' condition on channel Skinny/[EMAIL PROTECTED]Collected digit: [1] -- Asked to indicate 'Stop tone' condition on channel Skinny/
[EMAIL PROTECTED] -- Asked to indicate 'Stop tone' condition on channel Skinny/[EMAIL PROTECTED]Skinny [EMAIL PROTECTED] went on hookSkinny([EMAIL PROTECTED]): waitfordigit returned < 0skinny_hangup(Skinny/[EMAIL PROTECTED]
) on [EMAIL PROTECTED]the "Asked to indicate..." message after repeats indefinitely untilthe phone is hung up, and x-lite never sees the call.i'm running asterisk 1.2.1 (debian testing package) - below are a few
related sections of my config. my apologies if i've omittedsomething - this is my first experience with asterisk.thanks!-ben--sip.conf:[general]context=homebindport=5060bindaddr=
0.0.0.0srvlookup=yes--skinny.conf:[general]port = 2000bindaddr = 0.0.0.0dateFormat = Y-M-DkeepAlive = 120[office]device=SEP00D0BA03AB66
host=192.168.1.50context=homeline => 1234model=12SPversion=P00203010003callerid="office" <84>--extensions.conf:[general]static=yes
writeprotect=noautofallthrough=yesclearglobalvars=nopriorityjumping=no[globals]CONSOLE=Console/dspIAXINFO=guestTRUNK=Zap/g2TRUNKMSD=1[home]exten => 81,1,Dial(SIP/ion,20,tr)
exten => 82,1,Dial(SIP/quark,20,tr)exten => 83,Dial(SIP/proton,20,tr)exten => 84,1,Dial(Skinny/[EMAIL PROTECTED],20,tr)___--Bandwidth and Colocation provided by
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Re: [Asterisk-Users] Preferred editor(s) dialplan coding?
Bill Gibbs wrote: Vim for everything -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pete Barnwell Sent: Friday, March 03, 2006 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Preferred editor(s) dialplan coding? Emacs... On Sat, 2006-03-04 at 01:35 +0100, adibar wrote: >Vim forever ;-) > >On Fri, Mar 03, 2006 at 03:06:02PM -0500, S McGowan wrote: > emacs for me :) -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] SIP Problem - Asterisk to Provider Gateway
On Mar 3, 2006, at 1:46 PM, Gavin Adams wrote: Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN -> SIP Provider -> SIP -> * but outgoing calls are not. Call setup takes place and the caller can hear about 1-2 seconds of audio before the SIP provider cancels the call and sends back a BYE message. They haven't made any changes on their end (metaswitch). [snip] Okay, by changing the sip.conf entry to an IP address instead of a / etc/host entry has resolved the problem. I'll do further research next week to see if it's * or the remote SIP gateway choking on the entry. Regards, --- Gavin ___ --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] Upgrading to 1.2.5?
I think you need to: - pull the 1.2.5 tar.gz file from ftp/http - extract it into a dir (tar xvfz filename) - cd into it - excecute make upgrade and make install Cheers, John >>> [EMAIL PROTECTED] 04-03-06 21:12 >>> Probably just me being dumb, but I am trying to update my asterisk to the latest (1.2.5) When I go to my /usr/src/asterisk I type: make upgrade make install This seems to be doing it's thing, but when I type show version from the console (after a restart) I still get: Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a Power Macintosh running Darwin on 2006-03-04 20:48:08 UTC This seems like the same version number I had before also the copyright only goes through 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
[Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel
Hi all, I hope everyone is doing well. I just joined the list, and I've really enjoyed all I have read about asterisk so far. Unfortunately, I'm having a bit of trouble implementing this thing :). By the way ... I did my best to search the forums, and also to use google extensively, and while I have found pages with people with the same problem, ... The fix suggested on those sites, didn't work for me. Here's what I have: Results of uname -r: 2.6.9-22.0.2.106.unsupportedsmp Arch: X86_64 If you need more specs on the machine or OS, please let me know. I downloaded and have been following the asterisk book, and in chapter three I followed all the instructions on downloading the sources, untarring them, and so forth. Zaptel compiled without a hitch, as did the rest of the asterisk packages. I modified udev, and I restarted the box: ... I did: /etc/init.d/zaptel start I get: Loading zaptel framework: FATAL: Module zaptel not found. [FAILED] Waiting for zap to come online...Error: missing /dev/zap! If I do /sbin/modprobe zaptel I get: FATAL: Module zaptel not found. If I do /sbin/modprobe ztdummy I get: FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy Also, if i run: /etc/init.d/zaptel reload I get: Reloading ztcfg: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected [FAILED] If I go back to /usr/src/zaptel-1.2.4 and I do make ztdummy I get: cc ztdummy.o -o ztdummy /usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21): In function `_start': : undefined reference to `main' ztdummy.o(.text+0xc): In function `ztdummy_timer': /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined reference to `zt_receive' ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155: undefined reference t o `zt_transmit' ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156: undefined reference t o `jiffies' ztdummy.o(.text+0x4d): In function `init_module': include/linux/slab.h:93: undefined reference to `malloc_sizes' ztdummy.o(.text+0x52):include/linux/slab.h:93: undefined reference to `kmem_cach e_alloc' ztdummy.o(.text+0x6a): In function `init_module': /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined reference to `printk' ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192: undefined reference to `zt_register' ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239: undefined reference to `printk' ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240: undefined reference to `kfree' ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261: undefined reference to `jiffies' ztdummy.o(.text+0x23d): In function `init_module': include/linux/timer.h:87: undefined reference to `__mod_timer' ztdummy.o(.text+0x255): In function `init_module': /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined reference to `printk' ztdummy.o(.text+0x27c): In function `cleanup_module': /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined reference to `del_timer' ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303: undefined reference to `zt_unregister' ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304: undefined reference to `kfree' ztdummy.o(.text+0x39): In function `ztdummy_timer': include/linux/timer.h:87: undefined reference to `__mod_timer' ztdummy.o(.text+0x2b0): In function `cleanup_module': /usr/src/zaptel-1.2.4/ztdummy.c:310: undefined reference to `printk' ztdummy.o(__param+0x10): undefined reference to `param_set_int' ztdummy.o(__param+0x18): undefined reference to `param_get_int' collect2: ld returned 1 exit status make: *** [ztdummy] Error 1 Any ideas? I know I posted things in some wrong order here, but when I actually did them as a part of the install progress: I followed the order layed out in chapter 3 of the book. Thanks for any assistance. Take care, Sina ___ --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] Upgrading to 1.2.5?
Martin Joseph wrote: Probably just me being dumb, but I am trying to update my asterisk to the latest (1.2.5) When I go to my /usr/src/asterisk I type: make upgrade make install This seems to be doing it's thing, but when I type show version from the console (after a restart) I still get: Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a Power Macintosh running Darwin on 2006-03-04 20:48:08 UTC This seems like the same version number I had before also the copyright only goes through 2005? Thanks for your help. Marty Marty, Try "make update" and "make upgrade". -- Kristian Kielhofner ___ --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] Upgrading to 1.2.5?
Probably just me being dumb, but I am trying to update my asterisk to the latest (1.2.5) When I go to my /usr/src/asterisk I type: make upgrade make install This seems to be doing it's thing, but when I type show version from the console (after a restart) I still get: Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a Power Macintosh running Darwin on 2006-03-04 20:48:08 UTC This seems like the same version number I had before also the copyright only goes through 2005? Thanks for your help. Marty ___ --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] incoming calls dropout on PRI over TE110p
I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta, drops all calls once a day for 1 second, calls getting stuck, quite unpleasant! I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul C Sent: Wednesday, 1 March 2006 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p > Paul C wrote: >> I am running Asterisk 1.0.9 and have been running all my calls through a >> VSP over a IAX2 trunk however we have recently purchased and connected a >> TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make >> outgoing calls via it fine, however incoming calls are dropped after a >> few seconds ( or as soon as a command like Playback, or the call is >> picked up if forwarded to a SIP extensions ). >> SNIP << > > overlapdial should usually be no in my experience. Okay I've turned that to no with no change. I've just got off the phone to Optus and apparently they had a client in melbourne last week and they fixed the problem by turning crc checking off at the optus end. I don't suppose that was anybody on 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
[Asterisk-Users] # (send immediately) and dialplan broken on PAP2?
We have a bunch of PAP2s, and using the # to send immediately does not work as described in the manual. The PAP still waits for the "Interdigit_Short_Timer" to expire before sending the dial string. In addition, the dialplan does not cause the string to be sent immediately as it should. Here's the dialplan I'm using: (*x.|xxx|[3469]11|[2-9]xxS0|1[2-9]xx[2-9]xxS0|.) We've seen this behavior with both firmware 3.1.3 and 3.1.9. Has anyone else experienced this? Thanks, Barton ___ --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] really need help with outgoing calls..PSTN errors
At 10:03 PM 03/03/2006, you wrote:
You mean like this
exten => ww_9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
thanks
More likely:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})
Ira
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Re: [Asterisk-Users] really need help with outgoing calls..PSTN errors
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _9XX,2,Congestion()
exten => _9XX,102,Congestion()
I think these 3 lines need to have a 1 added like this:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/1${EXTEN:1})
exten => _9XX,2,Congestion()
exten => _9XX,102,Congestion()
Looks like they are intended to take long distance without a 1 which
is OK as long as you have * pass it on for you.
Ira
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RE: [Asterisk-Users] snom 320 MWI light
> Someone urged us to implement this behavior. I guess there was a large > company that told us that they were not able to send another MWI that > indicates that the messages were deleted... So far people could live > with this smart idea (it was not our idea). I don't understand why you have to be "urged" to implement this behaviour. As I see it, all the other SIP phones I have talking to my * box keep the MWI light on until they receive a message from * telling them otherwise. The snom's on the other hand turn off the MWI immediately when calling the VM number. This is incorrect behaviour to the best of my knowledge. Nabeel ___ --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 1.2.5 Released
On Mar 4, 2006, at 7:54 AM, The Asterisk Development Team wrote: Asterisk 1.2.5 is now available for download on the ftp. See the ChangeLog for details about what has changed. Reading the changelog I notice the following... I suppose it should say incorrect? 2006-02-17 01:55 + [r10301-10368] Russell Bryant <[EMAIL PROTECTED]> * jitterbuf.c: fix incorrent index calculation for jitterbuffer history (issue #6517) ___ --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] help with asterisk installation
Hello, You should install (with YAST2) termcap. And also mpg123, which is not included in the distro... Otherwise, you can simply install asterisk from Yast2 directly (but older version: 1.0.9.4). Alban Le Samedi 4 Mars 2006 18:55, Pete Barnwell a écrit : > On Sat, 2006-03-04 at 13:04 +, (pg) Zeeshan wrote: > > Dear All, > > > > I am new to both linux and asterisk. i want to install asterisk on suse > > 10 but receiving the error: "termcap support not found". i dont know what > > to do? i shall be highly obliged if someone helps me. > > > > zeeshan > > Are you installing from source, rpm or what (also which version of > asterisk)? What did you do to get that error message? Post more detail > and somebody might be able to help. > > Cheers > > Pete > > ___ > --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] Program Buttons on Cisco 79xx Phones
The only two programmable buttons are the 'Messages' and 'Services' and 'Directory buttons'. They are all configured in sip_default: messages_uri: "number to dial" services_url: "xml file to load" directory_url: "xml file to load" Cheers, Omar On 3/4/06, Kevin Steil <[EMAIL PROTECTED]> wrote: > > > > Does anyone have a good resource to learn how to program the soft and hard > buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware…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 > > > ___ --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] help with asterisk installation
On Sat, 2006-03-04 at 13:04 +, (pg) Zeeshan wrote: > Dear All, > > I am new to both linux and asterisk. i want to install asterisk on suse 10 > but receiving the error: "termcap support not found". i dont know what to do? > i shall be highly obliged if someone helps me. > > zeeshan Are you installing from source, rpm or what (also which version of asterisk)? What did you do to get that error message? Post more detail and somebody might be able to help. Cheers Pete ___ --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] snom 320 MWI light
Someone urged us to implement this behavior. I guess there was a large company that told us that they were not able to send another MWI that indicates that the messages were deleted... So far people could live with this smart idea (it was not our idea). CS (yes I am from snom) > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Nabeel Jafferali > Sent: Friday, March 03, 2006 9:01 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] snom 320 MWI light > > > I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the > > sip.conf entry, I have [EMAIL PROTECTED] and vmexten=*98. > > > > The light on the snom 320 turns on when I have voicemail and the > > retrieve button dials the correct extensions. > > > > However, the light turns off immediately after making the call to > > voicemail, even if I do not check the voicemail. > > FYI Received the following from a vendor: > > Currently there is not a way to keep the MWI light to stay on > after hitting retrieve button on the Snom. The best option > at this point is to set > checkmwi=1 in the general section of your sip.conf file. > This will cause the light to turn back on shortly if there > are un-checked messages waiting. > > ___ > --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] A room full of Cisco 7960s behind NAT
The problem is the remote server. Asterisk is able to drop the media stream and allow the SIP phones to communicate directly, which has both its drawbacks and advantages depending on how you plan to use asterisk. For this to take place you'll need the planets to be in the proper alignment and the following scenerio: 1.) the clients need to agree on a set of codecs so asterisk doesn't have to transcode them. 2.) both clients configured as 'canreinvite=yes' and 'nat=no' 3.) asterisk doens't have to listen for additional DTMF tones Since your phones are behind a nat using a remote asterisk server the calls will always have to route through the * box even if you were calling an associate in the cube next to you. If you were to install a local asterisk box it could handle this problem and also connect to the remote server as well. So a call between two SIP phones will have to go through the remote server? Or can those two phones be aware of each other? ___ --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: Asterisk-Users Digest, Vol 20, Issue 20
Message: 6 Date: Fri, 03 Mar 2006 17:32:47 + From: Conrad Wood <[EMAIL PROTECTED]@[EMAIL PROTECTED]@conradwood.net> Subject: Re: [Asterisk-Users] Problem with NAT!!! To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain On Fri, 2006-03-03 at 10:45 +0100, serge messa wrote: > Hi all > >I'm a newbie in asterisk.I install asterisk server > successfully. I configure this server to traverse NAT. > Using Xlite clients, i make a call between 2 local > networks through Internet.Asterisk server is > installed on a host with public IP. client A (in the > LAn A) and client B (in the LAN B) are registered. > When i make a call from the LAN A to the LAn B, > everything goes well.But, when i try to make a call > from the Lan B to the Lan A, the xlite client B, How do connect Lan A and Lan B to the internet? Do they both have a public static IP or are they dynamically assigned? Are they both the same routers? It might be far off but here's a couple of possible reasons: a) either one LAN keeps changing it's public IP (or just bad timing that the IP of Lan A changed when you tried to place your call to it) b) Either Router (a or b) might not allow the relevant packets through to xlite (or to the internet) Can you give more details on your configuration? Can you provide asterisk logs? Conrad Lan A and Lan B are separate by Internet. My server has a static public Ip. The router of the Lan A and the router of the Lan B have statics public adresses. When i make a call from a host with local ip inside the Lan A to the client on a host with a local ip inside the Lan B, every things goes well, but, when i make a call from, a lan B to the Lan A, my xlite client display connecting, and after a time, it display time out 408, call failed. That's the problem. I don't know how i can fix this problem. Serge ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.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] Re: Asterisk Question
> I actually got it all working - but it's great to see where we did the > same > thing, and where we differ. > > I ended up using the 'pop' perl command - inside a loop to go back one > item > at a time through my list > > PaulH Nice work! Perl = TMTOWTDI = There's More Than One Way To Do It -MC ___ --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] (no subject)
HiDoes someone have a better sql query for selecting the provider used by LCDial application than the one proposed in the tgz ? It's far from working well with most of price lists.I tried to tweak it somehow with more or less success.Regards, Michel -- Michel Luczak[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] Asterisk 1.2.5 Released
Asterisk 1.2.5 is now available for download on the ftp. See the ChangeLog for details about what has changed. ftp://ftp.digium.com/pub/telephony/asterisk/ As mentioned in the release announcement for Zaptel 1.2.4, our releases now contain some extra files. The Asterisk release is available as asterisk-1.2.5.tar.gz. However, there is also a patch against the previous release as an option for a smaller download, asterisk-1.2.5-patch.gz. For both the release tarballs and release patches, we have provided SHA-1 sums and PGP signatures. To verify the releases, you will need the public keys of both [EMAIL PROTECTED] and [EMAIL PROTECTED] Both are available on the keyserver, pgp.mit.edu. Thank you for your continued support of Asterisk! -- The Asterisk Development Team ___ --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] Changing caller id on transfer
My dial plan is as simple as it gets: exten => 101,1,Dial(sip/sip101,180,Ttr) But I'm doing blind transfers and you're doing attended transfers. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Dinesh Nair > Sent: Saturday, March 04, 2006 7:05 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Changing caller id on transfer > > > > On 03/03/06 04:17 Cosmin Prund said the following: > > How can I change the caller id on a transferred call so the called party > > knows the call has been transferred from a colleague and it's not coming > > directly from our outside lines? > > ironic ! we're trying to do the reverse: > > 1. call comes in via our digium zap lines > 2. receptionist answers > 3. receptionist uses atxfer (*1 in features.conf) to transfer to extension > 4. called extension sees callerid of receptionist's extension > > we'd like #4 to read, "extension called extension sees callerid of > original > caller" ! > > could you post your dialplan ? > > -- > Regards, /\_/\ "All dogs go to heaven." > [EMAIL PROTECTED](0 0)http://www.alphaque.com/ > +==oOO--(_)--OOo > ==+ > | for a in past present future; do > | > | for b in clients employers associates relatives neighbours pets; do > | > | echo "The opinions here in no way reflect the opinions of my $a $b." > | > | done; done > | > += > + > ___ > --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] Changing caller id on transfer
Thanks for the tip!
I shoud have found this on my own...
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of C F
> Sent: Friday, March 03, 2006 5:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Changing caller id on transfer
>
> Use the following variable in the dialplan to figure out that it has
> been transfered (this only works on a blind transfer) and change CID
> as you wish:
> # ${BLINDTRANSFER}: The active SIP channel that dialed the number.
> This will return the SIP Channel that dialed the number when doing
> blind transfers - see BLINDTRANSFER
> This is a paste from:
> http://www.voip-info.org/wiki-asterisk+variables
> and is also in:
> /doc/README.variables and
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[Asterisk-Users] Asterisk to a Huawei softX3000
Greetings, I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be something as simple as perhaps the register string which is currently @ although I've tried a number of variations without success. Here's a snipit from sip.conf allow=ulawauth=md5disallow=alldtmf=inbandhost=xxx.xxx.xxx.xxxinsecure=verysecret=xxxtype=peerusername=xxx Has anyone been able to register Asterisk against this Huawei switch? Normally I'd just muddle though it but I've spent the day working on this with NO success. I should also mention I've done ethereal dumps of devices that successfully register and I can't spot any differences. Thanks! Glen ___ --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] *** Yet another boring weekend? Test new Asterisk features in development!
In Sweden, where I live, it's snowing like crazy. The Stockholm area is covered in white stuff and there's really no reason to leave the computer and get out anywhere. More white stuff is coming down all the time. Boring. I am sure your weekend is no better - rain, snow or just another boring sunny day. Let's find something cool to do during this weekend! Join the cool crowd that tests the test branch during evenings and weekends. The dudes and dudettes that proudly contributes by reporting everything from simple spelling errors to crashes and strange noices from their Asterisk boxes. The people who knows what is going on in the Asterisk development circles - the Asterisk Test Team! I've updated the test branch to the latest version of my SIP peermatch code. This is quite a large code change, but not as large a functional change. However, it changes some basic functionality: * The sip_user structure is gone * Incoming calls are matched first by user from: name, then peer From: name, then IP. * Friends are now *one* in-memory object. In most cases, this means you can change type=friend to type=peer for local phones on the same LAN. This will also improve SIP subscriptions (blinking lights) and call limits, since for both friends and peers, we now have *one* object in memory that handles the limit for both incoming and outgoing calls. During the week, I've also added a few other patches by other contributors. Read the README.test-this-branch here: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/ README.test-this-branch?view=markup ** PLEASE help the community, please test this branch. Check it out like this svn checkout http://svn.digium.com/svn/asterisk/team/oej/test-this- branch test-trunk Then cd into test-trunk and run "make" then "make install" Report any bugs in the proper open bug in the bug tracker. If you like new functions, add a comment that this works for you. Provide feedback, make our work easier. Run "svn update" from time to time to get the latest version. Any changes from trunk will be merged into this code. Read the README.test-this-branch file to get more information. Thank you for your help! /Olle PS. Obviously, this is test code, not recommended to be closer than 2 miles (20 kilometers) from your production servers. ___ --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] really need help with outgoing calls..PSTN errors
Have you tried dialing an 800 number? Does that work? This extension:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
seems to be missing one X since it's only 10 digits long. Your PSTN
probably requires a 1 to be dialed also. On the other hand,
exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
you should probably be matching this extension instead although you
won't be able to match anywhere that has an area code that starts with
an 8 or 9. (905, 916, 914 as a few examples).
MARK.
sdgesa gaeharth wrote:
I cant seem to get outgoing calls to be placed properly .. No matter
what I try I get an error from the PSTN company saying that the "call
can not be completed as dialed" or "you need to dial a one..." The
asterisk debugging seems to show the correct number being dialed out
of the zap interface... the "9" is being stripped and there is a "1"
where it is supposed to be. I am thinking it is a problem between the
zap interface and the PSTN.
thanks
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
ATTENDANT=1001
OUTBOUNDTRUNK=ZAP/g1
[extentions]
exten => _10XX,1,Ringing
exten => _10XX,2,Dial(SIP/${EXTEN},20)
exten => _10XX,3,Answer
exten => _10XX,4,VoiceMail([EMAIL PROTECTED]
)
exten => _10XX,5,Hangup
[voicemail]
exten => _910XX,1,Wait(1)
exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
[local]
include => extentions
include => voicemail
[incoming]
exten => s,1,Answer
exten => s,n,Wait(2)
exten => s,n,Set(TIMEOUT(response)=15)
exten => s,n,Background(company-intro)
exten => s,n,WaitExten()
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup()
exten => 0,1,Dial(SIP/${ATTENDANT},20)
exten => 1,1,Directory(voicemail,extentions,f)
exten => 2,1,Directory(voicemail,extentions)
exten => 1234,1,Playback(abandon-all-hope)
include => extentions
exten => i,1,Playback(vm-goodbye)
exten => i,2,Hangup()
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
[outbound]
ignorepat => 9
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _9XX,2,Congestion()
exten => _9XX,102,Congestion()
exten => _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _91800NXX,2,Congestion()
exten => _91800NXX,102,Congestion()
exten => _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _91888NXX,2,Congestion()
exten => _91888NXX,102,Congestion()
exten => _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _91877NXX,2,Congestion()
exten => _91877NXX,102,Congestion()
exten => _91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _91866NXX,2,Congestion()
exten => _91866NXX,102,Congestion()
exten => _91900NXX,1,Congestion()
exten => _91976NXX,1,Congestion()
exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _91[1234567]XXNXX,2,Congestion()
exten => _91[1234567]XXNXX,102,Congestion()
exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411)
exten => 0,1,Dial(${OUTBOUNDTRUNK}/0)
[local-access]
include => local
include => outbound
zapata.conf:
[channels]
group => 1
language=en
context=incoming
signalling=fxs_ks
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callerid => "Dulles Micro, LLC" <703 450 5000>
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
channel => 1
zaptel.conf:
fxsks=1,2,3,4
loadzone = us
defaultzone=us
Brings words and photos together (easily) with
PhotoMail
- it's free and works with Yahoo! Mail.
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Re: [Asterisk-Users] Child PID's
On Thu, Mar 02, 2006 at 02:19:29PM -0600, Matt Schulte wrote: > All, I'm not sure how to word this question but we're noticing a lot of > our asterisk boxes no longer have multiple asterisk child processes. > i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used > to seeing 8+ .. There is no rhyme or reason to it, and we're using the > safe_asterisk script which has always worked in the past. Ast 1.2.4, zap > 1.2.4, naturally.. > > All my research has revealed nothing, regarding this, any suggestions? > What I'm worried about, of course, is the single process getting > overloaded with CPU calls and potentially denying service. A single process, just as before. Multiple theads of it, as before. Take a look at /proc//tasks Also, try 'ps auxm' rather than 'ps aux' . See ps(1) and look for "threads". -- 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] MixMonitor Problems -- sssshh, don't be too loud
On 3/3/06, Gary Richardson <[EMAIL PROTECTED]> wrote: I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811.. I'm running 1.2.1 and most of mine get cut short too. I posted this on the list a few months ago and nobody had any suggestions. BJ said I should probably post a bug on it but I haven't had time to continue to troubleshoot it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been watching change logs and hadn't seen anything surrounding mixmonitor so I've let it go. Please continue to update us if anyone gets some resolution. I'm glad to know there are lots of us experiencing this. That should be the catalyst to get it fixed. -Brian ___ --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] Program Buttons on Cisco 79xx Phones
Duh. Thanks. I spent all my time looking for SIP XML configs that my brain is fried now. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Saturday, March 04, 2006 8:14 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones On 07:39, Sat 04 Mar 06, Bill Gibbs wrote: > Awesome. > Any URLs to the XML templates for all the features? > The SCCP firmware doesn't appear to have any in the zip file. Have a look here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services good luck -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info 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?" ___ --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] Program Buttons on Cisco 79xx Phones
On 07:39, Sat 04 Mar 06, Bill Gibbs wrote: > Awesome. > Any URLs to the XML templates for all the features? > The SCCP firmware doesn't appear to have any in the zip file. Have a look here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services good luck -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info 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?" ___ --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] help with asterisk installation
Dear All, I am new to both linux and asterisk. i want to install asterisk on suse 10 but receiving the error: "termcap support not found". i dont know what to do? i shall be highly obliged if someone helps me. zeeshan ___ --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] Two PBX
> > HELLO everyone > > I am having two alcatel 4600 digital phone PBXs .. They are situated in > two > locations 15km apart. > > I want users or extension in both PBXs to be able to dial and receive > calls > from each others through those 30 channels in the E1 .. > > I have line of sight so i am planing to use a wireless link between these > two. Still i need a gateway or convertor from the PBXs to ip or lan ... > Can > i do this using two asterisk pc and two E1 card provided that the acatel > has > an E1 port in it . > > Is that possible to do this link ?? Can i make asterisk pcs transperant ?? > > What is the simplest configuration to make ??? > > > Hi, yes, this can be done. Ans yes, Asterisk will be 'transperant'. So you put an Asterisk box next to each alcatel connected over an E1.So what I am thinking is each of the alcatel's has a route on it to send numbers of the other branch over the link to the Asterisk boxes. In Asterisk you have a dialplan that whatever number you receive over the E1, because you know it it a number of the other branch(), you just dial the IP of the other Asterisk, and in this Asterisk whatever number you recieve you just dial over the E1 to the alcatel. I hope I have not over simplified it :) yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Program Buttons on Cisco 79xx Phones
Awesome. Any URLs to the XML templates for all the features? The SCCP firmware doesn't appear to have any in the zip file. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Saturday, March 04, 2006 6:07 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones On 10:34, Sat 04 Mar 06, Ron Wellsted wrote: > The SIP firmware does not allow the softkeys to be programmed :( > > Unfortunately you have to make a choice: > SIP firmware - Easy to implement on *, but poor XML support > SCCP firmware - poor/non-trivial asterisk support, great XML support. I have to correct you here. using chan_sccp gives you the same, and for some models more, functionality then CCM. Look here: http://chan-sccp.org I use it for both Cisco phones as kirk dect phones and it works great. And yes, the XML support is so much better then the SIP firmware. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info 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?" ___ --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] Two PBX
I have a similar configuration: two Alcatel PCX 4400 with E1+DID and a dialplan shared between sites. How do you plan to configure Asterisk boxes to share dialplan? DUNDI? Thanks for any info From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hafez AzzamSent: Friday, March 03, 2006 8:56 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Two PBX HELLO everyone I am having two alcatel 4600 digital phone PBXs .. They are situated in two locations 15km apart. I want users or extension in both PBXs to be able to dial and receive calls from each others through those 30 channels in the E1 .. I have line of sight so i am planing to use a wireless link between these two. Still i need a gateway or convertor from the PBXs to ip or lan ... Can i do this using two asterisk pc and two E1 card provided that the acatel has an E1 port in it . Is that possible to do this link ?? Can i make asterisk pcs transperant ?? What is the simplest configuration to make ??? Help regards to all Hazirliksiz yakalanmamak için MSN hava durumu hizmetinizde! Burayi tiklayin! ___ --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] What hardware to use for ISDN in Romania
The line is supposed to be standard EUROISDN; I found no mention of DSS1 on the technical specs page, I only found EUROISDN. OK, it should be DSS1 then ... The TELCO is going to provide me with a NT equipment that has two analog ports and two S0 ports; Of the two S0 ports one is supposed to be used to connect the PBX to the NT; I've got no idea what the other ports are, I can only guess the two analog ports will give me access to the two voice channels using plain-old analog phones. Good, then you get such an ISDN-TA already from your TELCO. Usually here in Germany, you just get an NT with the S0 interface and all other behind is your choice (incl. buying and setting up an TA ...) ... But your TELCO seems to avoid these user problems. The S0 ports should be in parallel. Normally, you can connect up to 8 devices to them but can use only 2 channels at the same time. I'm asking about "risks" because I ran through the wiki's and ended up very confused because it seems Asterisk's support for ISDN is driver-dependent and drivers are obviously kind of hardware dependent. Absolutely right ... The "risk" I'm talking about is signing up for a ISDN contract only to find I can't get the drivers going, or I can't fully use the service. Since I don't have access to any other ISDN installation OR ISDN hardware, all I've got to go on is email, google and the wiki! As a matter of fact I don't know what hardware to look for! Do I buy this from a telco provider or from a computer hardware shop? Am I looking for something listed in the "modem" category or for some other hardware? Since there aren't that many Asterisk consultants in Romania I don't really know where to ask. And yes, I did find MODULO in Bucharest (listed on the wiki as consultants) but they did not return my last two emails so I'm on my own :-) Uups, well. Then there is some risk ... Are you able to compile a new kernel for an ASTERISK installation, if that would be needed for a proper hardware support ? The ISDN-hardware (PC-card) you can easily get at EBAY or in a PC shop. It should run as "PCI-ISDN card (with HFC-S chipset)". For an easy start and good performance I personally would recommend to take a AVM B1 PCI (active ISDN card, allows faxing as well), get the recent driver sources for the helper applications from AVM and use chan_capi-cm an ASTERISK channel driver. At least here (even with older AVM B1 ISA cards), that runs without problems under 2.6.14.xx kernel and ASTERISK 1.2.x ... But as you might have read here the last days depending on your TELCO and its signalization of calls (example AUSTRIA) there can problems appear which nobody could think of ... Regards, Jürgen > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Juergen K. Zick > Sent: Saturday, March 04, 2006 11:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] What hardware to use for ISDN in Romania > > HI There > > if the line is a standard EUROISDN with DSS1 protocol then it's not a risk > at all to take and to connect it. You get an S0 interface from your TELCO > and there cou can plug in any EUROISDN compliant equipment e.g. TAs, > phones > and, of course, ASTERISK ... > > I would suggest that you AT LEAST try to get an ISDN-TA (ISDN <-->PSTN > converter for "old" analogue phones and fax machines) as well, for testing > and backup purposes. They are cheap to get e.g. for abt 2-5 EUR e.g. on > EBAY. > > Depending on your budgets (time and money), experiences and skills you can > equip your ASTERISK box with incoming and outgoing ISDN channels. You will > find quite a lot config examples for that, supposingly ISDN-cards with > HFC-S chipsets are the most versatile ... However, ISDN drivers are still > a > bit tricky, but youo have depending on your kernel version at least > ISDN4LINUX, vISDN, mISDN and chan_modem, chan_capi, chan_capi-cm, > chan_misdn as config options ... > > Anything else you shoul dbe able to find in the WiKis ... > > Regards, > > Jürgen > > > > >My land-line provider (Romtelecom) has a very nice offer for ISDN. All in > >all they offer me a digital land-line with 1 base number + 2 MSN's and > that > >would make a grate addition to my full-time home office. > > > >Romtelecom say they're providing EURO-ISDN and the line is compatible > with > >any euro-isdn compliant equipment. They say they'll install a NT at my > >office and this NT will provide me with 1 (one) SO (or was that S0 - zero > >opposed to the letter O?) port to connect to my PBX. > > > >My questions: > >What hardware do I use to connect the line to my Asterisk? > >What are the risks involved (bad drivers etc)? > >Has any one used this in Romania? > > > >Thanks! > > > >___ > >--Bandwidth and Colocation provided by Easynews.com -- > > > >Asterisk-Users mailing list > >To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones
On 10:34, Sat 04 Mar 06, Ron Wellsted wrote: > The SIP firmware does not allow the softkeys to be programmed :( > > Unfortunately you have to make a choice: > SIP firmware - Easy to implement on *, but poor XML support > SCCP firmware - poor/non-trivial asterisk support, great XML support. I have to correct you here. using chan_sccp gives you the same, and for some models more, functionality then CCM. Look here: http://chan-sccp.org I use it for both Cisco phones as kirk dect phones and it works great. And yes, the XML support is so much better then the SIP firmware. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info 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?" ___ --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] A room full of Cisco 7960s behind NAT
On 03:10, Sat 04 Mar 06, Tzafrir Cohen wrote: > On Tue, Feb 28, 2006 at 05:25:40PM -0700, Damon Estep wrote: > > Try nat=yes and qualify=yes in sip.conf. > > So a call between two SIP phones will have to go through the remote > server? Or can those two phones be aware of each other? Yes. But without this things will not work. What you can do is put a local asterisk in there to do the routing of internal numbers so reinvites work again. Or you can use something like SER to do the outbound routing -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info 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?" ___ --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] Program Buttons on Cisco 79xx Phones
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bill Gibbs wrote: > All I have found was stuff about softkey templates in Call Manager. If > there is any programming we could do without CM that would be > fantastic!! For some reason I can?t get an iDivert key to show up on my > 7940G! > The SIP firmware does not allow the softkeys to be programmed :( Unfortunately you have to make a choice: SIP firmware - Easy to implement on *, but poor XML support SCCP firmware - poor/non-trivial asterisk support, great XML support. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBRAltQktP/KMNOfRbAQL0Igf7BC21bznYmTEviyjXD5cALBL1zDDfi1gl k9YNXi4Guasf33UiPwFi290SfPtjZpE9Striae7U1rmoxspbl+WeL/Yyi0mOyxwD GtaYGgA0TixqblrPN52DUZnAvAfZ/AW0CjtQxOjoUJ9Y4YwI+YeeVSQa63e6BUvp tfg9e2VHmP4ml8AQ1CsDRSCM8Rw+L28WZWWnIgA61piBr/G0KuQ2gekIC6AZhuNO zIhGU4MB0YaDJP0sR4za6tgoSnA+5xfOuo4nRn953sF9wtjYlyyoeTc3fZqFd4/B zPEW6va4t98lFMBN0rMtJpVVl3nHEmX6TVotAqehix57gbvdsVB62Q== =/StB -END PGP SIGNATURE- ___ --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] What hardware to use for ISDN in Romania
The line is supposed to be standard EUROISDN; I found no mention of DSS1 on the technical specs page, I only found EUROISDN. The TELCO is going to provide me with a NT equipment that has two analog ports and two S0 ports; Of the two S0 ports one is supposed to be used to connect the PBX to the NT; I've got no idea what the other ports are, I can only guess the two analog ports will give me access to the two voice channels using plain-old analog phones. I'm asking about "risks" because I ran through the wiki's and ended up very confused because it seems Asterisk's support for ISDN is driver-dependent and drivers are obviously kind of hardware dependent. The "risk" I'm talking about is signing up for a ISDN contract only to find I can't get the drivers going, or I can't fully use the service. Since I don't have access to any other ISDN installation OR ISDN hardware, all I've got to go on is email, google and the wiki! As a matter of fact I don't know what hardware to look for! Do I buy this from a telco provider or from a computer hardware shop? Am I looking for something listed in the "modem" category or for some other hardware? Since there aren't that many Asterisk consultants in Romania I don't really know where to ask. And yes, I did find MODULO in Bucharest (listed on the wiki as consultants) but they did not return my last two emails so I'm on my own :-) > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Juergen K. Zick > Sent: Saturday, March 04, 2006 11:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] What hardware to use for ISDN in Romania > > HI There > > if the line is a standard EUROISDN with DSS1 protocol then it's not a risk > at all to take and to connect it. You get an S0 interface from your TELCO > and there cou can plug in any EUROISDN compliant equipment e.g. TAs, > phones > and, of course, ASTERISK ... > > I would suggest that you AT LEAST try to get an ISDN-TA (ISDN <-->PSTN > converter for "old" analogue phones and fax machines) as well, for testing > and backup purposes. They are cheap to get e.g. for abt 2-5 EUR e.g. on > EBAY. > > Depending on your budgets (time and money), experiences and skills you can > equip your ASTERISK box with incoming and outgoing ISDN channels. You will > find quite a lot config examples for that, supposingly ISDN-cards with > HFC-S chipsets are the most versatile ... However, ISDN drivers are still > a > bit tricky, but youo have depending on your kernel version at least > ISDN4LINUX, vISDN, mISDN and chan_modem, chan_capi, chan_capi-cm, > chan_misdn as config options ... > > Anything else you shoul dbe able to find in the WiKis ... > > Regards, > > Jürgen > > > > >My land-line provider (Romtelecom) has a very nice offer for ISDN. All in > >all they offer me a digital land-line with 1 base number + 2 MSN's and > that > >would make a grate addition to my full-time home office. > > > >Romtelecom say they're providing EURO-ISDN and the line is compatible > with > >any euro-isdn compliant equipment. They say they'll install a NT at my > >office and this NT will provide me with 1 (one) SO (or was that S0 - zero > >opposed to the letter O?) port to connect to my PBX. > > > >My questions: > >What hardware do I use to connect the line to my Asterisk? > >What are the risks involved (bad drivers etc)? > >Has any one used this in Romania? > > > >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 > > ___ > --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] Authenticated SIP NOtify with 1.2.4?
I have been working with authenticated notifys for auto resync my autoprovisined devices. But it seems to stop the state machine, and when Endpoint answers 401 Unauthorized, the Sip Notify command from cli, does not answer with a Authenticated Notify? Have i misconfigured something? Regards ___ --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] What hardware to use for ISDN in Romania
HI There if the line is a standard EUROISDN with DSS1 protocol then it's not a risk at all to take and to connect it. You get an S0 interface from your TELCO and there cou can plug in any EUROISDN compliant equipment e.g. TAs, phones and, of course, ASTERISK ... I would suggest that you AT LEAST try to get an ISDN-TA (ISDN <-->PSTN converter for "old" analogue phones and fax machines) as well, for testing and backup purposes. They are cheap to get e.g. for abt 2-5 EUR e.g. on EBAY. Depending on your budgets (time and money), experiences and skills you can equip your ASTERISK box with incoming and outgoing ISDN channels. You will find quite a lot config examples for that, supposingly ISDN-cards with HFC-S chipsets are the most versatile ... However, ISDN drivers are still a bit tricky, but youo have depending on your kernel version at least ISDN4LINUX, vISDN, mISDN and chan_modem, chan_capi, chan_capi-cm, chan_misdn as config options ... Anything else you shoul dbe able to find in the WiKis ... Regards, Jürgen My land-line provider (Romtelecom) has a very nice offer for ISDN. All in all they offer me a digital land-line with 1 base number + 2 MSN's and that would make a grate addition to my full-time home office. Romtelecom say they're providing EURO-ISDN and the line is compatible with any euro-isdn compliant equipment. They say they'll install a NT at my office and this NT will provide me with 1 (one) SO (or was that S0 - zero opposed to the letter O?) port to connect to my PBX. My questions: What hardware do I use to connect the line to my Asterisk? What are the risks involved (bad drivers etc)? Has any one used this in Romania? 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 ___ --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] Two PBX
HELLO everyone I am having two alcatel 4600 digital phone PBXs .. They are situated in two locations 15km apart.I want users or extension in both PBXs to be able to dial and receive calls from each others through those 30 channels in the E1 ..I have line of sight so i am planing to use a wireless link between these two. Still i need a gateway or convertor from the PBXs to ip or lan ... Can i do this using two asterisk pc and two E1 card&nb sp;provided that the acatel has an E1 port in it .Is that possible to do this link ?? Can i make asterisk pcs transperant ?? What is the simplest configuration to make ???Help regards to all Messenger sohbeti ile sesinizi, kendinizi ve duygularinizi ifade edin! Burayi tiklayin! ___ --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] Accept Unregistered GK Calls
Hi everyone, Could any tell me How i can accept unregistered Gatekeepers calls to my Asterisk Box? My customer is using another Gatekeeper and he want to use my Asterisk as a gateway for him to terminate the call using SIP protocol. and his Gatekeeper is not supported as end point to register my Asterisk Box. Here is waht i did the configuration but getting error: Error : "SIP/2.0 404 Not Found" sif.conf [from-SIPGK] type=friend host=cutomer_SIP_GK_IP_Address port=5060 nat=yes qualify=yes context=ivr-bal disallow=all allow=g729 extentions.con [ivr-bal] ;exten => _x.,1,Answer exten => _x.,2,AGI(ivr-bal.pl) Where ivr-bal.pl file is having very semple gsm file to play some voice. I will be appricate for your replies/ Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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
[Asterisk-Users] asterisk 1.2.5 cannot call a zap channel extension
Hi. I am using 1.2.5 and I have an extension using a zap fxs channel on a 400P Digium card. Now when thatextension is dialed with a timeout of 20 seconds it rings for about half a second and then the log says noone picked on after 2 seconds and so it goes to voicemail. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] What hardware to use for ISDN in Romania
Hello everyone. My land-line provider (Romtelecom) has a very nice offer for ISDN. All in all they offer me a digital land-line with 1 base number + 2 MSN's and that would make a grate addition to my full-time home office. Romtelecom say they're providing EURO-ISDN and the line is compatible with any euro-isdn compliant equipment. They say they'll install a NT at my office and this NT will provide me with 1 (one) SO (or was that S0 - zero opposed to the letter O?) port to connect to my PBX. My questions: What hardware do I use to connect the line to my Asterisk? What are the risks involved (bad drivers etc)? Has any one used this in Romania? 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
Re: [Asterisk-Users] Child PID's
On 4 Mar 2006, at 08:30, Paul Hewlett wrote: On Thursday 02 March 2006 22:19, Matt Schulte wrote: All, I'm not sure how to word this question but we're noticing a lot of our asterisk boxes no longer have multiple asterisk child processes. i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used to seeing 8+ .. There is no rhyme or reason to it, and we're using the safe_asterisk script which has always worked in the past. Ast 1.2.4, zap 1.2.4, naturally.. All my research has revealed nothing, regarding this, any suggestions? What I'm worried about, of course, is the single process getting overloaded with CPU calls and potentially denying service. Matt @ NetLogic Matt On 2.4 kernels you would be using the LinuxThreads implementation of POSIX threads. This emulated the POSIX threading model with some limitations - signals could only be delivered to the master thread - there was a 'hidden' control thread etc... Context switching performance in LinuxThreads was known to be poor (sometimes measured in whole seconds) IBM and RedHat worked on solving this problem - IBM's effort (NGPT) was abandoned in favour of REdHat's (NPTL). NPTL was introduced in RedHat 9 (as I remember). Most modern distro's now use NPTL and one can tell this by doing 'ps ax' and seeing only one asterisk task instead of many. Asterisk without NPTL is probably not good for high thruput sites. And Gentoo users beware - only 2006.0 has adopted NPTL as default. Previously one had to rebuild the toolchain by specifying USE flags "nptl nptlonly" to get NPTL threads (and this takes a long time...) Have u upgraded to a more modern distro recently ? Also take a look at the value of LD_ASSUME_KERNEL in your shell startup scripts and profiles. I had a similar weirdness caused by a line in my personal .profile LD_ASSUME_KERNEL=2.4.1 which I'd put in ages ago to test some (non asterisk) software and never taken out. [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
Re: [Asterisk-Users] test call quality
amaury BOSSE wrote: > Is there a free linux tool which can test voip call quality between two > Asterisk PBX. > > It will help me to test the WAN network between them. > > I have only found commercials ones, so if you know a free one, let me > know. For packet loss, rtt etc and a phone call check out: http://www.sineapps.com/sinestatiax.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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: G729 and Meetme
Martin Joseph wrote: > > On Mar 2, 2006, at 3:46 PM, Wai Wu wrote: > >> You can really mix G729 encoded frames. So I would guess that licenses >> are not needed for non-G279 devices. BTW, there is a difference >> conference app (forgot the name) that only mixes the two parties that >> have the loudest volumn. It sounds more efficent to me this way. There >> is no reason to listen to three or more party talking at the same time >> anyway. >> > I wish this was a joke. Sick and wrong is all I can say. :D Nah, iaxclient.sf.net has app_conference which does exactly that :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Child PID's
On Thursday 02 March 2006 22:19, Matt Schulte wrote: > All, I'm not sure how to word this question but we're noticing a lot of > our asterisk boxes no longer have multiple asterisk child processes. > i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used > to seeing 8+ .. There is no rhyme or reason to it, and we're using the > safe_asterisk script which has always worked in the past. Ast 1.2.4, zap > 1.2.4, naturally.. > > All my research has revealed nothing, regarding this, any suggestions? > What I'm worried about, of course, is the single process getting > overloaded with CPU calls and potentially denying service. > > Matt @ NetLogic Matt On 2.4 kernels you would be using the LinuxThreads implementation of POSIX threads. This emulated the POSIX threading model with some limitations - signals could only be delivered to the master thread - there was a 'hidden' control thread etc... Context switching performance in LinuxThreads was known to be poor (sometimes measured in whole seconds) IBM and RedHat worked on solving this problem - IBM's effort (NGPT) was abandoned in favour of REdHat's (NPTL). NPTL was introduced in RedHat 9 (as I remember). Most modern distro's now use NPTL and one can tell this by doing 'ps ax' and seeing only one asterisk task instead of many. Asterisk without NPTL is probably not good for high thruput sites. And Gentoo users beware - only 2006.0 has adopted NPTL as default. Previously one had to rebuild the toolchain by specifying USE flags "nptl nptlonly" to get NPTL threads (and this takes a long time...) Have u upgraded to a more modern distro recently ? Paul ___ --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 management interface
On Thu, Mar 02, 2006 at 02:32:43PM -0600, Anton Krall wrote: > |Try this: > | http://www.bicomsystems.com/docs/pbxware/ > > Looks very nice.. Is it GPL, GNU? Nither the GPL nor any other free software license. google for "pbxware". -- 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] IAX Video and Meetme
Hagen Rode wrote: > Hi > > I'm browsing around the internet looking for signs that the IAX client > library and app_meetme support video. > > I stumbled across this post by SteveK on the 27th of Feb 2006. > > "My company is looking to hire a full-time developer, who will be working > about 25-50% of the time on iaxclient; in particular to finally integrate, > build, polish and enhance video in iaxclient, add video support to > app_conference (also in iaxclient's CVS repository), and generally improve > the iaxclient audio and codebase." > > So my guess from this is that there is currently no support for video in > app_meetme, but that in the (hopefully not too distant) future, > app_conference will be the replacement for app_meetme and will have video > support. Replacement is a big word. I would expect that meetme in its current format will not be able to support video multiplexing. App_conference on the other hand looks like it may do. If you're into hacking code, the TIPIC libraries will support simple video communications at the moment, although I have been unable to compile it successfully. There was some discussion a while ago on the dev list regarding the replacement of meetme with app_conference, and the general consensus was that it wouldn't happen. This means that you will need to add it (possibly even in the future) to the apps directory and patch/alter the Makefile. OT: Mail me offlist if you are interested in building 3G/UMTS support into Asterisk. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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
