[asterisk-users] uk tole-free dids?
hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing Hardware
On Mon, 29 Sep 2008, Jim Boykin wrote: Thanks Gordon Mike for the response. What accuracy are you getting from zaptest/dahdi_test (and system info). Two more questions: 1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel. 2) What about CPU load? I run a custom compiled kernel with the high resolution timers HPET. CPU load for me is next to nothing. zttest for me: # zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% ^C --- Results after 17 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.996410 System was just carrying a few SIP - IAX calls at that point. Gordon Thanks Jim On Mon, Sep 29, 2008 at 5:02 AM, Mike Trest [EMAIL PROTECTED] wrote: Go for it. ztdummy is not an issue. I have used ztdummy with 220 simultaneous participants in 18 different conference groups. At one time, I had 60 machines running simultaneously in a FARM all of which were carrying the same 18 conference groups with over 200 participants active on each machine. ..mike.. At 11:23 AM 9/28/2008, Gordon Henderson wrote: On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI defunct processes + GSM Playback - HELP!
Hello. I've just installed asterisk-1.4.21.2 zaptel-1.4.12.1 chan_ss7-1.0.10 libpri-1.4.7 I am using Sangoma A104 card with wanpipe-3.2.7.1 drivers. My OS: Ubuntu 8.04 Server Kernel: 2.6.24-16-server I am getting a choppy GSM playback and too many defunct AGI processes when channel closes. i am using Perl or PHP, also 'agi-test.agi' going to defunct too... I was able to playback GSM files and running AGI with older version of Asterisk and Zaptel very well. I've just upgraded my servers to latest versions but it's too buggy now. Any ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
On Sun, 28 Sep 2008, Babcock, Michael Alex wrote: hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. As far as call charges are concerned, there is no such thing as local or long-distance in the UK now, and hadn't been for many years. We now have geographic and non geographic numbers. (Non geographics are sometimes called NGNs) The Geographics start 01 or 02 and are what they imply - mapped to a particular geographic area. (or are supposed to as some are ported into VoIP and moved) These numbers are relatively low cost to call and often can be called for free out of inclusive minutes or other deals with your phone company. NGNs include freephone numbers (aka toll-free in the US) - starting 0800, 0808 or 0500. NGNs can also be revenue generating - they are usually free to assign and maintain, but cost the caller more than calling a Geographic number and the terminating operator generates revenue from the call. These numbers typically start 084 or 087. The 087's generating more revenue than 084, and cost more to the caller. (There are other ranges of premium numbers - 09 and 070, but we'll not go there) So - If you want your friends to call you for free, then why not just allocate then a SIP account on your own system, then get them to run a softphone, or buy a hard-phone and then they can connect up? Failing that, they can get a SIP account with a multitude of UK (and European) based operators and make calls - sometimes for free, sometimes for a small per minute charge. And failing all that - because they don't have/want technology, you can get an account with one of these operators and terminate it in your own SIP/IAX hardware, then they can call you for the price of a standard UK geographic number - which may be free for them, depending on the package they have with their phone company. A final thing to note, there is a new range of non-geographic numbers in the UK which start 03. These cost the same to call as 01 and 02 numbers (or should), so you could allocate one of these and have it terminated on your own SIP/IAX account. People to do it via (other than me - I can't take US currency, but drop an email if interested anyway) in no particular order: www.gradwell.com, www.voiptalk.co.uk, www.voipon.co.uk, www.voip.co.uk, www.aql.co.uk and a few others. Hope this helps! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Creating Asterisk Binary Package
Is there a script to create an Asterisk binary package after it is compiled on one system. We do not want to compile Asterisk of each system where we want to run. I am sure there is a way but I could find it. Thanks Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
On Mon, Sep 29, 2008 at 03:41:26AM -0400, Jim Boykin wrote: Is there a script to create an Asterisk binary package after it is compiled on one system. We do not want to compile Asterisk of each system where we want to run. I am sure there is a way but I could find it. What system, specifically? With what libraries installed? Which of them are installed on the target system? rpm, deb and such automate much of the process. Chances are that there is already an existing such binary package for your distribution. Just grab it, fix it, and rebuild (and fix, and rebuild, and deploy, and fix and rebuild, etc.) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
Just copy the src folder and do `make install` on each machine? Then tar and copy the /etc/asterisk folder if config is important too. On 29 Sep 2008, at 08:41, Jim Boykin wrote: Is there a script to create an Asterisk binary package after it is compiled on one system. We do not want to compile Asterisk of each system where we want to run. I am sure there is a way but I could find it. Thanks Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA for large networks
Hi, I would like to know if someone can suggest a multi-port ATA worth buying (at least 8 ports). I have around 380 analog phones to convert to SIP extensions. So I need quite a few ATAs but they need to be enterprise-grade, ie. they need to be reliable and stable. I bought and have been using 11 Grandstream GXW4008 (8-port FXS ATA) in a production environment and have been experiencing stability and quality issues which are not acceptable in a large company. I chose Grandstream because: - it was a cheap way to start - I thought their products were stable and reliable because I had already heard their brand name So since my experience with 11 Grandstream GXW4008 has been overall negative (I need to reboot the devices too often!), I'd like to know if someone could help me decide what brand/model to buy. I would also need to find these products in Europe (or at least deliverable there). I've been considering a few products but I don't know if they are reliable: TopGate TG8048 (48 FXS) Soundwin S2400 (24 FXS) In other words, I'd really appreciate feedback from voip administrators (not from resellers) who have had experience testing their devices and are happy with them. Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] identify/find a channel to pick it up
hello, i want to build a pickup extension in a multiuser system, which means there are several different numbers which same extensions and so on. the phonenumbers are identified only with their ID in the database like +123- where after the - is the extension. the normal pickup function works with [EMAIL PROTECTED] but the problem i have is that i dont know the extension. so i could read out every possible extension from the db and build a pickupstring, but i think this would kill the system (if there are more than 200 extension on a number). this is for a global pickup function to pickup the next ringing channel for that number. the other way is that i could find the right channel, so i want to search for every channel which have the state ringing and the extension +123-... or is there something like a wildcard i can use for pickup like +123* sorry for my bad english, i hope its understandable what i mean. best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
Steven Howes wrote: Just copy the src folder and do `make install` on each machine? Then tar and copy the /etc/asterisk folder if config is important too. On 29 Sep 2008, at 08:41, Jim Boykin wrote: Is there a script to create an Asterisk binary package after it is compiled on one system. We do not want to compile Asterisk of each system where we want to run. I am sure there is a way but I could find it. Another way is after running ./configure --prefix=/your_prefered_layout and make, when running the make install command set the DESTDIR prefix to something like ~/asterisk and it will install everything under that prefix. e.g $ make DESTDIR=$HOME/asterisk install You can then make a tarball of the hierarchy from within the DESTDIR root and extract it into the right place (i.e. /) of any other host. Of course all this assumes: 1. you know what you are doing ;-) 2. your hosts are all using the same versions of kernels/libraries etc... HTH Alan Thanks Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
Philip Prindeville wrote:
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI:
${CALLERID(ani)})
exten = 208345,n,Goto(redfish-pstn,s,1)
...
[redfish-pstn]
exten = s,1(incoming),Noop()
exten = s,n,Answer()
exten = s,n,Wait(0.5)
...
some filters for bogus ANI's like 8 goes to badani below
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,nWaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
exten = s,n(badani),Log(DEBUG,ANI: ${CALLERID(ani)} clearing)
exten = s,n,Playback(privacy-unident)
exten = s,n,Wait(0.5)
exten = s,n,Congestion()
exten = s,n,Hangup()
include = redfish-extens
exten = i,1,NoOp(Invalid: ${EXTEN})
exten = i,n,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[redfish-extens]
...
exten = 113,1,Monitor(wav,,w); for debugging
exten = 113,n,Macro(stdexten,113,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
...
exten = 113,1,Macro(stdexten,119,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
Err, sorry. Typo. That was:
exten = 119,1,Macro(stdexten,119,${GUEST},redfish)
exten = 119,n,Goto(s,exten)
-Philip
So I don't get this at all. If I dial 208345, then enter '119' as
the extension, it rings on a few phones (including a Xlite softphone)
and if I pick up on any of those, I get one-way voice (I can hear the
caller but they can't hear me).
If I enter '113' as the extension, it rings on two SPA-942's (one of
which is the same as above, just a different line presentation)... and
if I answer, then I get two-way voice! Only difference is the Monitor()
statement.
I'm starting to suspect it's a CODEC issue in Asterisk, though (a) why
Asterisk would need to transcode a call between two uLaw endpoints, I
don't know... and (b) why is it staying in the Media path at all?
I have the SIP peer that the calls come in on as:
[sip-proxy]
...
type=peer
nat=no
canreinvite=no
reinvite=no
Anyone know why the Monitor() would change the duplex(ity) of the audio
stream? I'm baffled (no pun intended). And is there any debugging I
can turn on to reveal CODEC behavior that might differ from 113 and 119?
Thanks,
-Philip
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Re: [asterisk-users] ATA for large networks
Why not swap it all with just IP phone? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: Monday, September 29, 2008 4:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ATA for large networks Hi, I would like to know if someone can suggest a multi-port ATA worth buying (at least 8 ports). I have around 380 analog phones to convert to SIP extensions. So I need quite a few ATAs but they need to be enterprise-grade, ie. they need to be reliable and stable. I bought and have been using 11 Grandstream GXW4008 (8-port FXS ATA) in a production environment and have been experiencing stability and quality issues which are not acceptable in a large company. I chose Grandstream because: - it was a cheap way to start - I thought their products were stable and reliable because I had already heard their brand name So since my experience with 11 Grandstream GXW4008 has been overall negative (I need to reboot the devices too often!), I'd like to know if someone could help me decide what brand/model to buy. I would also need to find these products in Europe (or at least deliverable there). I've been considering a few products but I don't know if they are reliable: TopGate TG8048 (48 FXS) Soundwin S2400 (24 FXS) In other words, I'd really appreciate feedback from voip administrators (not from resellers) who have had experience testing their devices and are happy with them. Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
--- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... Thanks for the feedback, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Knowing incoming call technology and channel
Hi,
I've read www.voip-info.org but couldn't find the answer I'm after.
In diaplan, how can you know the technology and channel of an incoming call
?
I was thinking of something like :
[incoming]
exten = _,1,Set(CALLERID(num)=00${CALLERIDNUM})
exten = _,2,NoOp(This call comes from ${CHANNELTYPE})
exten = _,3,NoOp(This call comes from ${CHANNELNO})
exten = _,4,NoOp(Said differently this call comes from
${CHANNELTYPE}/${CHANNELNO})
My ultimate goal is to have this working with Zaptel channels (from a
bristuffed Asterisk).
Regards
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Re: [asterisk-users] Knowing incoming call technology and channel
Try this:
exten = _,1,Set(THISTECH=${CUT(CHANNEL,/,1)})
exten = _,n,NoOp(Technology is ${THISTECH})
exten = _,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)})
exten = _,n,NoOp(Channel is ${THISCHANNEL})
Olivier wrote:
Hi,
I've read www.voip-info.org http://www.voip-info.org but couldn't find
the answer I'm after.
In diaplan, how can you know the technology and channel of an incoming
call ?
I was thinking of something like :
[incoming]
exten = _,1,Set(CALLERID(num)=00${CALLERIDNUM})
exten = _,2,NoOp(This call comes from ${CHANNELTYPE})
exten = _,3,NoOp(This call comes from ${CHANNELNO})
exten = _,4,NoOp(Said differently this call comes from
${CHANNELTYPE}/${CHANNELNO})
My ultimate goal is to have this working with Zaptel channels (from a
bristuffed Asterisk).
Regards
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Re: [asterisk-users] Creating Asterisk Binary Package
On Mon, Sep 29, 2008 at 08:56:29AM +0100, Steven Howes wrote: Just copy the src folder and do `make install` on each machine? Then tar and copy the /etc/asterisk folder if config is important too. Self promotion If you want a self contained Asterisk environment, complete with logging directory, modules directory, configuration directory and whatever, and even with a wrapper script called asterisk, look at http://bugs.digium.com/11680 for live_ast /Self promotion But all of this does not but you independence from library dependencies. Do you have h323 installed? snmp? Zaptel? Binary packagees are a well-known problem, and one that has pretty good solutions. It's sad that the Asterisk hard-cores like re-inventing the wheel here. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
Thanks Alan, I will try it out. Seems like a solution. Your assumption is right, all system are same (ghosted). I am also looking at pre-build RPM and reusing their specs file. Anyone have input for building asterisk RPM. Thanks Jim On Mon, Sep 29, 2008 at 3:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 29, 2008 at 08:56:29AM +0100, Steven Howes wrote: Just copy the src folder and do `make install` on each machine? Then tar and copy the /etc/asterisk folder if config is important too. Self promotion If you want a self contained Asterisk environment, complete with logging directory, modules directory, configuration directory and whatever, and even with a wrapper script called asterisk, look at http://bugs.digium.com/11680 for live_ast /Self promotion But all of this does not but you independence from library dependencies. Do you have h323 installed? snmp? Zaptel? Binary packagees are a well-known problem, and one that has pretty good solutions. It's sad that the Asterisk hard-cores like re-inventing the wheel here. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
On Mon, Sep 29, 2008 at 03:34:45PM +0530, Jim Boykin wrote: Thanks Alan, I will try it out. Seems like a solution. Your assumption is right, all system are same (ghosted). I am also looking at pre-build RPM and reusing their specs file. Anyone have input for building asterisk RPM. Again, what distribution is it? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
2008/9/29 Alex Balashov [EMAIL PROTECTED]
Try this:
exten = _,1,Set(THISTECH=${CUT(CHANNEL,/,1)})
exten = _,n,NoOp(Technology is ${THISTECH})
exten = _,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)})
exten = _,n,NoOp(Channel is ${THISCHANNEL})
Hi,
I don't have any spare zaptel enabled system I could try this on, but I was
not aware of this CHANNEL variable.
Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables
Maybe, I will add a line in www.voip-info.org to keep others (me?) from
searching again.
Thanks for helping
Olivier wrote:
Hi,
I've read www.voip-info.org http://www.voip-info.org but couldn't find
the answer I'm after.
In diaplan, how can you know the technology and channel of an incoming
call ?
I was thinking of something like :
[incoming]
exten = _,1,Set(CALLERID(num)=00${CALLERIDNUM})
exten = _,2,NoOp(This call comes from ${CHANNELTYPE})
exten = _,3,NoOp(This call comes from ${CHANNELNO})
exten = _,4,NoOp(Said differently this call comes from
${CHANNELTYPE}/${CHANNELNO})
My ultimate goal is to have this working with Zaptel channels (from a
bristuffed Asterisk).
Regards
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Re: [asterisk-users] Creating Asterisk Binary Package
We use RHEL5, FC6, CentOS5. I will be happy to hear your inputs for any distribution you know. On Mon, Sep 29, 2008 at 3:41 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 29, 2008 at 03:34:45PM +0530, Jim Boykin wrote: Thanks Alan, I will try it out. Seems like a solution. Your assumption is right, all system are same (ghosted). I am also looking at pre-build RPM and reusing their specs file. Anyone have input for building asterisk RPM. Again, what distribution is it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disable CDR?
Hello
I'm running Asterisk 1.4.21.2 on FreeBSD 6.3.
This part of extensions.conf...
;play a menu, and expect user to type any extension 1-4 or 9
exten = s,n,Wait(1)
exten = s,n,Background(main_menu)
exten = s,n,WaitExten(5)
exten = s,n,Hangup()
exten = _[1-49],1,AGI(convert_app.phpcli|${EXTEN})
... triggers this message:
-- Executing [EMAIL PROTECTED]:5] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:6] BackGround(Zap/1-1, main_menu) in
new stack
-- Zap/1-1 Playing 'main_menu' (language 'fr')
== CDR updated on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] AGI(Zap/1-1,
convert_app.phpcli|1) in new stack
I don't use CDR. Provided this will not have dire consequences, how
can I disable this?
Thank you.
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Re: [asterisk-users] Creating Asterisk Binary Package
On Mon, Sep 29, 2008 at 03:51:35PM +0530, Jim Boykin wrote: We use RHEL5, FC6, CentOS5. I will be happy to hear your inputs for any distribution you know. Fedora 9 has a package, but I think it is asterisk 1.6.0-rc9. Some SRPMs of lesser quality for Centos 5: http://yum.trixbox.org/centos/5/SRPMS/ http://yum.trixbox.org/centos/5/SRPMS/repodata/repoview/A.group.html http://yum.trixbox.org/centos/5/SRPMS/repodata/repoview/asterisk-0-1.4.21.2-2.html http://repo.elastix.org/centos/5/updates/SRPMS/repodata/ http://repo.elastix.org/centos/5/updates/SRPMS/repodata/repoview/A.group.html http://repo.elastix.org/centos/5/updates/SRPMS/repodata/repoview/asterisk-1-1.4.21.2-3.html (Elastix's developers have this funny habbit of making the path leading to that directory non-indexed) The lesser quality shows e.g. in the fact that the changelog is not always updated. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
I know about those packages. Questions is how do we use those packages to build our own RPM. We use asterisk SVN trunk. Thanks Jim On Mon, Sep 29, 2008 at 4:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 29, 2008 at 03:51:35PM +0530, Jim Boykin wrote: We use RHEL5, FC6, CentOS5. I will be happy to hear your inputs for any distribution you know. Fedora 9 has a package, but I think it is asterisk 1.6.0-rc9. Some SRPMs of lesser quality for Centos 5: http://yum.trixbox.org/centos/5/SRPMS/ http://yum.trixbox.org/centos/5/SRPMS/repodata/repoview/A.group.html http://yum.trixbox.org/centos/5/SRPMS/repodata/repoview/asterisk-0-1.4.21.2-2.html http://repo.elastix.org/centos/5/updates/SRPMS/repodata/ http://repo.elastix.org/centos/5/updates/SRPMS/repodata/repoview/A.group.html http://repo.elastix.org/centos/5/updates/SRPMS/repodata/repoview/asterisk-1-1.4.21.2-3.html (Elastix's developers have this funny habbit of making the path leading to that directory non-indexed) The lesser quality shows e.g. in the fact that the changelog is not always updated. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
On Mon, Sep 29, 2008 at 12:00 PM, Vieri [EMAIL PROTECTED] wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... Thanks for the feedback, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can actually buy IP Phones that have an Ethernet switch incorporated in them and take advantage of your existing cabling infrastructure with no need for new switches etc. I have only used Cisco 188 ATA without any troubles at all but those only have 2 ports and not applicable for your situation. Otherwise you can try Software SIP Phones if everyone is using a PC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel
Olivier schrieb:
In diaplan, how can you know the technology and channel of an incoming call
?
${CHANNEL(channeltype)}
Philipp Kempgen
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[asterisk-users] OT - Avantages of ISDN PtP and PtmP
Hi, Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is the way to connect businesses but if you read http://public.swbell.net/ISDN/connect.html you would think the opposite. Can anyone elaborate a bit PtP or PtmP respective advantages ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
2008/9/29 Thanos Koukoulis [EMAIL PROTECTED] Otherwise you can try Software SIP Phones if everyone is using a PC Beside few people, it's very difficult to swap hardphones and softphones. Have you observed any successful experience in doing so ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/IAX Interworking ip CANCEL behavior
Hello, i have two asterisk boxes connected to each other via iax trunk. all is working fine, only if one extension cancles the call during ringing, the far end extension still rings three times until call is also terminated (both extensions are connected via SIP) this does not happen after successfull call setup, the sip BYE is processed correctly. Any ideas, what i forgot to configure ?! best regards, Andreas M. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Avantages of ISDN PtP and PtmP
On Mon, Sep 29, 2008 at 01:26:10PM +0200, Olivier wrote: Hi, Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is the way to connect businesses but if you read http://public.swbell.net/ISDN/connect.html you would think the opposite. Can anyone elaborate a bit PtP or PtmP respective advantages ? With analog phone lines you can have multiple handsets share the same physical line. That is: multiple FXSs and one FXS. Only one talk at a time. PtMP is an attempt to preserve that feature: multiple CPE units can share the same physical connection to a network unit. Only up to two talks at a time. Another nice feature of analog handsets is that they are powered from the FXS. Likewise BRI phones can be powered from the network unit. Those two features are quite nice when your equipment is a simple phone. They are mostly useless for a PBX: your PBX will have an independent power source anyway. And will most likely want to handle all the line by itself. The problem is that those features come with a price tag of complexity. For instance, many providers want to save power and hence drop the (even layer 1) connection to a ptmp bri cpe unit because it is probably an isdn phone that takes some precious power (some 25V, IIRC). As a result, your PBX cannot really tell if the red alert it has is faked or not. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Avantages of ISDN PtP and PtmP
Olivier schrieb: Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is the way to connect businesses but if you read http://public.swbell.net/ISDN/connect.html you would think the opposite. That's not true, although multipoint sounds better than just one point. :-) PtMP: Usually you connect your devices (phones etc.) directly to the line (although you could connect a PBX). Each of the phones has a totally different number. PtMP is what home users get unless they request something else. PtP: You connect just one device which is your PBX. You get a block of numbers (xx / xxx / / ...). The nice thing is that you can easily map these external DID numbers to internal extensions, i.e. ..xx - xx More expensive than PtMP. btw: _PRI_ is always PtP. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable CDR?
On Mon, Sep 29, 2008 at 1:25 PM, Vincent [EMAIL PROTECTED] wrote:
Hello
I'm running Asterisk 1.4.21.2 on FreeBSD 6.3.
This part of extensions.conf...
;play a menu, and expect user to type any extension 1-4 or 9
exten = s,n,Wait(1)
exten = s,n,Background(main_menu)
exten = s,n,WaitExten(5)
exten = s,n,Hangup()
exten = _[1-49],1,AGI(convert_app.phpcli|${EXTEN})
... triggers this message:
-- Executing [EMAIL PROTECTED]:5] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:6] BackGround(Zap/1-1, main_menu) in
new stack
-- Zap/1-1 Playing 'main_menu' (language 'fr')
== CDR updated on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] AGI(Zap/1-1,
convert_app.phpcli|1) in new stack
I don't use CDR. Provided this will not have dire consequences, how
can I disable this?
in cdr.conf:
[general]
enable=no
You may also unload CDR modules. For this do:
ast-dev14*CLI module show like cdr
Module Description
Use Count
cdr_manager.so Asterisk Manager Interface CDR Backend 0
cdr_custom.so Customizable Comma Separated Values CDR 0
app_forkcdr.so Fork The CDR into 2 separate entities0
app_cdr.so Tell Asterisk to not maintain a CDR for 0
app_setcdruserfield.so CDR user field apps 0
func_cdr.soCDR dialplan function0
cdr_addon_mysql.so MySQL CDR Backend0
7 modules loaded
And add in modules.conf:
noload = cdr_csv.so
noload = cdr_odbc.so
noload = cdr_pgsql.so
noload = cdr_sqlite.so
noload = cdr_sqlite3_custom.so
for each module not used.
Regards,
Atis
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VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] OT - Avantages of ISDN PtP and PtmP
2008/9/29 Olivier [EMAIL PROTECTED]: Hi, Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is the way to connect businesses but if you read http://public.swbell.net/ISDN/connect.html you would think the opposite. Can anyone elaborate a bit PtP or PtmP respective advantages ? IMHO: PtMP should only be used if you have Multiple devices connected to your ISDN (Hence the Multiple part in its name) - This setup means that for each outgoing call, the calling device has to negotiate almost from scratch, access to a B-channel, and every inbound call is sent (broadcast) to every connected device to give it the chance to grab it. This is an almost compeletly chaotic and stateless environment (I know, of course there IS state, just a lot less of it). PtP on the other hand is stateful - The one device negotiates a connection when it comes up, and monitors the line constantly, so it knows if (for example) the line goes down. Calls in both directions are sent to the one known endpoint, directly addressed to a B-channel that it already knows should be available. Also, if a call comes in to an unrecognised DDI/address, PtMP can only time-out (no-one grabs the call) where PtP can dynamically know that the call is rejected and handle it properly For 99% of Asterisk installs, where Asterisk is managing/routing all calls on a line, PtP is going to be the right choice. Cheers, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Avantages of ISDN PtP and PtmP
2008/9/29 Philipp Kempgen [EMAIL PROTECTED] Olivier schrieb: Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is the way to connect businesses but if you read http://public.swbell.net/ISDN/connect.html you would think the opposite. That's not true, although multipoint sounds better than just one point. :-) From http://public.swbell.net/ISDN/connect.html : If you only intend to connect a single device/application to your ISDN line, then you only need the point-to-point configuration. With the point-to-point configuration you are assigned a single phone number per ISDN line (not one for each B-channel). If you intend to connect multiple devices/applications, then you need the multipoint configuration. With multipoint configuration you are assigned a phone number for each device connected. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bizarre international call problem.
On Fri, 26 Sep 2008 15:04:30 -0400 (EDT), Ken D'Ambrosio wrote: So I'm confused: any ideas on how this worked when the PBX was hooked straight to the PSTN? Is there some SS7 signal or something that says, This is an international call, when the number has no 011 preface? I'd hate to have to revert, but I will if need be... *sigh* the provider may be tagging it on. have you checked pridialplan, or prilocaldialplan settings and playing around with that in zapata.conf ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... Thanks for the feedback, Vieri You'll probably want to use FXS channel banks rather than an ATA. At that kind of scale, I'd call Rhino or Xorcom and have them make the recommendation. You will still end up with a large number of devices and likely several asterisk servers to coordinate all of this. When you really look at the numbers, finding a way to use IP phones may not be that much more than the overall hardware cost involved to do this right with analog lines. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CLI and verbosity level
Hi, Whenever I'm logging in with asterisk -r command, I can see that the verbosity and debug levels are set to a value which is different from the last ones I left when I logged off from CLI. Where are those default levels defined ? I can't see any related option in logger.conf. Any hint ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI and verbosity level
Olivier wrote: Hi, Whenever I'm logging in with asterisk -r command, I can see that the verbosity and debug levels are set to a value which is different from the last ones I left when I logged off from CLI. Where are those default levels defined ? I can't see any related option in logger.conf. Any hint ? The verbosity will be at least as high as the last time you entered the CLI. For example if two times ago, you entered with 5 v's then entered the last time with 1 v, you will still be at 5 v's. You can change this behavior using: CLIcore set verbose X where X is the new level you want (2, 3 ...) Hope that makes sense. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX
Hello Cosmin,
I also tried this, and it doesn't work. I think it is a bug but i'm not
sure. Let us know if you find any solution.
Regards,
Serghei Gutanu
Cosmin Nistor wrote:
Hello and thank you for replyes.
Eric, I looked for it on the mailing list and google and
did not find something relevant to be 100% sure that this feature is
not supported.
Some information clare I founded in http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroupswhere
it says that for IAX channels I can use the pickup feature from
features.conf.
I was looking for an anser to understand if this is
supported or not, not to lose more time trying to make it work.
Shazaum , thank you for your anser, the application Pickup
works ok.
My problem is that this application issued from the dial-plan is
directed pickup, thos means that I have to know the exten that is
ringing.
I have difficulties because I an using call queues and the channel is
not anymore only the exten that is ringing, and if I want to pikup a
call that is comming from a queue, I cannot do this with app Pickup(at
least I did not find any way to do this--any help from somebody who did
is apreciated.)
Also, since IAX is developed by asterisk, is strange that
for SIP there is support, and for IAX, this kind of application is not
supported--this is why I asked, maybe I am doing something wrong.
In this case(if it is not supportted), shoul we/I open a
bug repot to Digium?
Botton line, what i am trying to do is to pickup any call
that cames in, direct call, transfered call, queue call, using IAX, and
I am wondering if this is possible in any way.
Regards,
Cosmin
I believe chan_iax2 does not support call pickup. Search the archives.
Shazaum wrote:
already tested with an exten?
ex:
exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _*8.,n,Hangup()
2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello list
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse
I want to put the users in call queues and I want anybody from the
company to be able to pick a ringing channel, even if is in a queue.
Whwn using Sip protocol for the users, everithing is going fine, I
can pickup any ringing extension from the group using *8.
But the problem appears when I am using IAX protocol. When issuing
*8 from the IAX phone, asterisk tryes to find the *8 in the dialling
rules returning:
*CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
http://10.0.0.30:4569
[Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
Rejected connect attempt from 10.0.0.30 http://10.0.0.30, request
'[EMAIL PROTECTED]' mailto:[EMAIL PROTECTED] does not exist
This I think is wrong, is something like asterisk cannot read from
features.
With the same setting, when using SIP, i get:
*CLI == Using SIP RTP CoS mark 5
[Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
handle_request_invite: Nothing to pick up for
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
and it works ok.
I am wondering if any had this problem before and if you can help me
figure it out(how to make it work--or if is a bug), or find a
sollution using the app pickup.
I tryed using asterisk 1.4.13, asterisk 1.4.21.2 http://1.4.21.2,
asterisk 1.6-rc6 and always the same problem ocurs.
Regards,
Cosmin
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Re: [asterisk-users] OT - Avantages of ISDN PtP and PtmP
2008/9/29 Olivier [EMAIL PROTECTED]: From http://public.swbell.net/ISDN/connect.html : If you only intend to connect a single device/application to your ISDN line, then you only need the point-to-point configuration. With the point-to-point configuration you are assigned a single phone number per ISDN line (not one for each B-channel). If you intend to connect multiple devices/applications, then you need the multipoint configuration. With multipoint configuration you are assigned a phone number for each device connected. That is utter rubbish. PtP mode is completely unrelated to how many numbers are allocated on the line. It is possible that 1-to-1 is a common implementation where an ISDN phone is being used, but that does not make it a requirement. A SETUP packet arrives and it contains the caller-ID (usually) and the called number (or a representation of it) - In PtMP, all devices that are interested in handling the called number then fight over who answers the call based in the received information, and in PtP, the one device decides what to do with the call whether it wanted it or not. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
In other words, I'd really appreciate feedback from voip administrators (not from resellers) who have had experience testing their devices and are happy with them. I would recommend the Linksys SPA8000 (8 port ATA). It is as solid and reliable as the SPA2102. Andres http://www.neuroredes.com Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
On Mon, Sep 29, 2008 at 2:30 PM, Olivier [EMAIL PROTECTED] wrote: 2008/9/29 Thanos Koukoulis [EMAIL PROTECTED] Otherwise you can try Software SIP Phones if everyone is using a PC Beside few people, it's very difficult to swap hardphones and softphones. Have you observed any successful experience in doing so ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well my experience with major softphone deployment was using Cisco Call Manager. In that situation we had ~150 softphones and ~100 IP Phones. Apart from the initial configuration and taking extra care that the machines running the softphones were configured correctly we experienced very few problems. Extra care had to be taken to select good quality headphones and USB headphones did tend to work better than ones connected to sound cards. Problems arose when the users 'tinkered' with their machines (which we eventually locked down pretty hard) or an unrelated application crashing causing the whole machine to slow down to a crawl and affecting voice quality significantly. Otherwise things did work rather smoothly and the extra mobility gains for some of our users (moving around the office with their mobile PCs) made it worthwhile. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
On Mon, 29 Sep 2008, Andres wrote: In other words, I'd really appreciate feedback from voip administrators (not from resellers) who have had experience testing their devices and are happy with them. I would recommend the Linksys SPA8000 (8 port ATA). It is as solid and reliable as the SPA2102. The OP has 300 phones. That's 38 SPA devices. And while you might think it's solid and reliable, I have one customer using 3 of them and they're not impressed with echo on their existing analog network. This is high-end channel bank territory. Multiple E1s - traditional channel banks, or something like multiple 24-port Xorcom units or the like... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI and verbosity level [SOLVED]
2008/9/29 Darrick Hartman [EMAIL PROTECTED] Olivier wrote: Hi, Whenever I'm logging in with asterisk -r command, I can see that the verbosity and debug levels are set to a value which is different from the last ones I left when I logged off from CLI. Where are those default levels defined ? I can't see any related option in logger.conf. Any hint ? The verbosity will be at least as high as the last time you entered the CLI. Do you mean the highest between last and previous last times ? For example if two times ago, you entered with 5 v's then entered the last time with 1 v, you will still be at 5 v's. You can change this behavior using: CLIcore set verbose X where X is the new level you want (2, 3 ...) Hope that makes sense. It does but whatever I tried, it defaulted to 5. Anyway, with grep in /etc/asterisk/*.conf, I found those 2 lines in asterisk.conf : debug=5 verbose=5 That should explain why I couldn't revert to a lower level of verbosity. Thanks for helping Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
I will second this opinion. This may be going a little off topic, but is there a way to lock the voip section of the ata so that the end user can not change settings in this area, but as far as the ip settings go and other sections the user will be able to access them? Thanks, Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: Monday, September 29, 2008 9:58 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ATA for large networks In other words, I'd really appreciate feedback from voip administrators (not from resellers) who have had experience testing their devices and are happy with them. I would recommend the Linksys SPA8000 (8 port ATA). It is as solid and reliable as the SPA2102. Andres http://www.neuroredes.com Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.5/1696 - Release Date: 9/29/2008 7:40 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bizarre international call problem.
the provider may be tagging it on. have you checked pridialplan, or prilocaldialplan settings and playing around with that in zapata.conf ? Oooh. That makes sense. I've poked around, but don't really see much documentation on this. 'Cause going outbound is easy, but how do I check to see if the inbound (from my legacy PBX) has tagged a given call as international? Thanks, again! -Ken -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Quintum Tenor AX. Just glance over the manual. The are far better and in my experience just as reliable as a channel bank. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:26 AM, Eric ManxPower Wieling [EMAIL PROTECTED]wrote: The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Tom Moore wrote: I will second this opinion. This may be going a little off topic, but is there a way to lock the voip section of the ata so that the end user can not change settings in this area, but as far as the ip settings go and other sections the user will be able to access them? Yes it is possible but a little bit more involved. First you setup the device to download the profile from a tftp or http server where you setup the xml file. The file will have all the parameters defined like this: Proxy_1_ ua=na192.168.1.200/Proxy_1_ What that ua=na mean is that the user will have No Access to the parameter. The other 2 options are ro and rw, read-only and read-write respectively. There is no way to define these directly on the SPA web page, you have to use a provisioning file in order to define these permissions. Andres http://www.neuroredes.com Thanks, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
the following two lines exist in the extensions_additional.conf [from-max]exten = _X,1,Answerexten = _X,n,Queue(8000,tr,,) and it DOES exist in the output of the 'show dialplan' [ Context 'from-max' created by 'pbx_config' ] '_X' = 1. Answer() [pbx_config]2. Queue(8000|tr||) [pbx_config] yet my system doesn't use it to route regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 28 Sep 2008 23:31:46 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Issues On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Channel Banks only On Mon, Sep 29, 2008 at 10:44 AM, Steve Totaro [EMAIL PROTECTED] wrote: Quintum Tenor AX. Just glance over the manual. The are far better and in my experience just as reliable as a channel bank. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:26 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Try AudioCodes MP-124 which is 24 ports FXS. I have one but haven't used it much yet, so I cannot comment about its quiality. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
Eric ManxPower Wieling wrote: Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. Web and/or context-searchable documentation will ALWAYS win out over a somewhat loose collection of text files. That's basic UI psychology 101. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel Lines - How many are in use..
Hello, I have a question. We have a 8 port FXO card in our asterisk server plugged into 8 analog lines. Is there a way to tell at how many of those ports are in use (AKA, actually on a call)? I tried zap show status and zap show channel [num] but I don't see anything that might be helpful. Singer -- *Singer X.J. Wang* /Systems Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Cell: (613) 218-9184 Fax:(613) 565-8710 Email: [EMAIL PROTECTED] MSN:[EMAIL PROTECTED] Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer email;internet:[EMAIL PROTECTED] tel;work:(613) 565-8696 x298 x-mozilla-html:TRUE version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
So, should we (I can do it, if desired) write a script that polls subversion docs directory and imports it into voip-info.org when the the docs are changed? I'd be glad to write and host such a script if the community desires the feature. -josiah SIP wrote: Eric ManxPower Wieling wrote: Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. Web and/or context-searchable documentation will ALWAYS win out over a somewhat loose collection of text files. That's basic UI psychology 101. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Lines - How many are in use..
I normally use 'core show channels' and check for 'Zap/' in the channel string. Are you trying to do it in an automated way, like from AGI? Singer Wang wrote: Hello, I have a question. We have a 8 port FXO card in our asterisk server plugged into 8 analog lines. Is there a way to tell at how many of those ports are in use (AKA, actually on a call)? I tried zap show status and zap show channel [num] but I don't see anything that might be helpful. Singer -- *Singer X.J. Wang* /Systems Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free:(877) 798-4426 x298 Cell: (613) 218-9184 Fax: (613) 565-8710 Email:[EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Yahoo:pythianwang AIM: pythianwang ICQ: 201253 Gadu-Gadu:6817795 Tencent QQ: 858310404 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Huh, please try to form a complete thought Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:53 AM, C F [EMAIL PROTECTED] wrote: Channel Banks only On Mon, Sep 29, 2008 at 10:44 AM, Steve Totaro [EMAIL PROTECTED] wrote: Quintum Tenor AX. Just glance over the manual. The are far better and in my experience just as reliable as a channel bank. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:26 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Lines - How many are in use..
Try with: core show channels verbose or core show channels concise Regards, Luis Morales On Tue, Sep 30, 2008 at 10:38 AM, Singer Wang [EMAIL PROTECTED] wrote: Hello, I have a question. We have a 8 port FXO card in our asterisk server plugged into 8 analog lines. Is there a way to tell at how many of those ports are in use (AKA, actually on a call)? I tried zap show status and zap show channel [num] but I don't see anything that might be helpful. Singer -- Singer X.J. Wang Systems Engineer The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Cell: (613) 218-9184 Fax: (613) 565-8710 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Yahoo: pythianwang AIM: pythianwang ICQ: 201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Source of SIP Remote host can't match request NOTIFY
Hi, I've got a lot of Remote host can't match request NOTIFY. I've read http://lists.digium.com/pipermail/asterisk-users/2008-May/210709.html . Recommendation is enable SIP debug to see what packets are causing this, and if it's voicemail notifications, turn them off in sip.conf How would you proceed to get those debug ? The exact message is : WARNING[2990]: chan_sip.c:12900 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. As this 192.168.52.31 adresse is my Asterisk server address, I don't have a clue to focus debugging on the device that replies 481 to Asterisk. Unfortunately, I've got several dozens of devices and all with QUALIFY=yes. So in 60 seconds, I get hundred of SIP debug lines. Is there a smart way to handle this ? Something like SIP DEBUG 481 ? Anyway, do you think it could make sense to ask (or develop) a more explicit WARNING message such as Remote host 192.168.52.123 can't match ... ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
I don't see why not, Voip-info is very outdated in most respects. Most of it with bad examples, dating to Asterisk 1.x era. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan Sent: September 29, 2008 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED] So, should we (I can do it, if desired) write a script that polls subversion docs directory and imports it into voip-info.org when the the docs are changed? I'd be glad to write and host such a script if the community desires the feature. -josiah SIP wrote: Eric ManxPower Wieling wrote: Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. Web and/or context-searchable documentation will ALWAYS win out over a somewhat loose collection of text files. That's basic UI psychology 101. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
what are 70 numbers? On Sep 28, 2008, at 10:26 PM, Geraint Lee wrote: You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
The Linksys SPA-8000 is an 8 port FXS unit that works very well. For that volume you should also consider either a Channelbank or maybe a Xorcom Astribank. You can get those in 24 or 32 port versions. On Mon, 2008-09-29 at 02:00 -0700, Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... Thanks for the feedback, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
thanks for all this information. michael On Sep 28, 2008, at 11:37 PM, Gordon Henderson wrote: On Sun, 28 Sep 2008, Babcock, Michael Alex wrote: hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. As far as call charges are concerned, there is no such thing as local or long-distance in the UK now, and hadn't been for many years. We now have geographic and non geographic numbers. (Non geographics are sometimes called NGNs) The Geographics start 01 or 02 and are what they imply - mapped to a particular geographic area. (or are supposed to as some are ported into VoIP and moved) These numbers are relatively low cost to call and often can be called for free out of inclusive minutes or other deals with your phone company. NGNs include freephone numbers (aka toll-free in the US) - starting 0800, 0808 or 0500. NGNs can also be revenue generating - they are usually free to assign and maintain, but cost the caller more than calling a Geographic number and the terminating operator generates revenue from the call. These numbers typically start 084 or 087. The 087's generating more revenue than 084, and cost more to the caller. (There are other ranges of premium numbers - 09 and 070, but we'll not go there) So - If you want your friends to call you for free, then why not just allocate then a SIP account on your own system, then get them to run a softphone, or buy a hard-phone and then they can connect up? Failing that, they can get a SIP account with a multitude of UK (and European) based operators and make calls - sometimes for free, sometimes for a small per minute charge. And failing all that - because they don't have/want technology, you can get an account with one of these operators and terminate it in your own SIP/IAX hardware, then they can call you for the price of a standard UK geographic number - which may be free for them, depending on the package they have with their phone company. A final thing to note, there is a new range of non-geographic numbers in the UK which start 03. These cost the same to call as 01 and 02 numbers (or should), so you could allocate one of these and have it terminated on your own SIP/IAX account. People to do it via (other than me - I can't take US currency, but drop an email if interested anyway) in no particular order: www.gradwell.com, www.voiptalk.co.uk, www.voipon.co.uk, www.voip.co.uk, www.aql.co.uk and a few others. Hope this helps! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]
You would want three pages, 1.2 docs, 1.4 docs, and 1.6 docs. Mark Hamilton wrote: I don't see why not, Voip-info is very outdated in most respects. Most of it with bad examples, dating to Asterisk 1.x era. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan Sent: September 29, 2008 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED] So, should we (I can do it, if desired) write a script that polls subversion docs directory and imports it into voip-info.org when the the docs are changed? I'd be glad to write and host such a script if the community desires the feature. -josiah SIP wrote: Eric ManxPower Wieling wrote: Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. Web and/or context-searchable documentation will ALWAYS win out over a somewhat loose collection of text files. That's basic UI psychology 101. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Lines - How many are in use..
We're not using AGI or anything like it. I'm scripting something that will be called from our trending tool (JFFNMS) to track usage of our analog lines. This way we have data when we ask for more money for more analog lines. Singer Josiah Bryan wrote: I normally use 'core show channels' and check for 'Zap/' in the channel string. Are you trying to do it in an automated way, like from AGI? Singer Wang wrote: Hello, I have a question. We have a 8 port FXO card in our asterisk server plugged into 8 analog lines. Is there a way to tell at how many of those ports are in use (AKA, actually on a call)? I tried zap show status and zap show channel [num] but I don't see anything that might be helpful. Singer -- *Singer X.J. Wang* /Systems Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Cell: (613) 218-9184 Fax: (613) 565-8710 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Yahoo: pythianwang AIM: pythianwang ICQ: 201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Singer X.J. Wang* /Systems Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Cell: (613) 218-9184 Fax:(613) 565-8710 Email: [EMAIL PROTECTED] MSN:[EMAIL PROTECTED] Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer email;internet:[EMAIL PROTECTED] tel;work:(613) 565-8696 x298 x-mozilla-html:TRUE version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
On Mon, 29 Sep 2008, Babcock, Michael Alex wrote: what are 70 numbers? Prefix 070 (then 8 more digits) These are so-called personal numbers. They're a blot and an anomaly. They are expensive to call and the recipient usually gets revenue from the calls. ie. they are premium rate, revenue generating numbers in disguise. In disguise becasue a lot of people (in the UK) don't realise this because they look like mobile numbers - which start 07[1-9] then 8 more digits, so they think they're calling a mobile, when in-fact it's costing them much more. Gordon On Sep 28, 2008, at 10:26 PM, Geraint Lee wrote: You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can Block a pri channel
Hi all, I'm new in astersik and like to know how can block a pri channel. It means I want block some channels on a pri link so nobody can occupy these channels, i.e channels 1 through 5 should be blocked. What is the Q931 message for blocking a channel? and can I see this message when I want use a protocol analyzer? thanks in advance Alireza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
right will stay away from them, smile. mike On Sep 29, 2008, at 9:01 AM, Gordon Henderson wrote: On Mon, 29 Sep 2008, Babcock, Michael Alex wrote: what are 70 numbers? Prefix 070 (then 8 more digits) These are so-called personal numbers. They're a blot and an anomaly. They are expensive to call and the recipient usually gets revenue from the calls. ie. they are premium rate, revenue generating numbers in disguise. In disguise becasue a lot of people (in the UK) don't realise this because they look like mobile numbers - which start 07[1-9] then 8 more digits, so they think they're calling a mobile, when in-fact it's costing them much more. Gordon On Sep 28, 2008, at 10:26 PM, Geraint Lee wrote: You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
On Mon, Sep 29, 2008 at 09:17:11AM -0800, Babcock, Michael Alex wrote: right will stay away from them, smile. On Mon, 29 Sep 2008, Babcock, Michael Alex wrote: what are 70 numbers? Prefix 070 (then 8 more digits) These are so-called personal numbers. They're a blot and an anomaly. They are expensive to call and the recipient usually gets revenue from the calls. ie. they are premium rate, revenue generating numbers in disguise. Worth noting that UK premium rate numbers are covered by PhonePayPlus (the regulator for PRS). http://www.phonepayplus.org.uk/consumers/faq/default.asp 070 MAY be covered if they are used for PRS services (not for personal numbering). Ofcom are intending to move PRS out of 070 (and want to make all 07 mobile). There are oddities of course, the Channel Islands adhere to UK numbering plans by agreement with the UK gov (or however it works) but Channel Island mobiles and landline numbers are treated as foreign calls even though they are in UK number space. Great isn't it. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cheap FXO Card?
I have many of the Intel PCI modems in the field working for some time, but I am trying to find a source for more of them. IMO places like x100p.com are a rip off -- $40 for a PCI modem? I recall getting the AMI modems a few years ago for $10. So does anyone know where I can find the PCI WinModem that is detected as X100P or X101P for a better price? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
On Mon, 2008-09-29 at 14:51 +, Tariq .. wrote: the following two lines exist in the extensions_additional.conf [from-max] exten = _X,1,Answer exten = _X,n,Queue(8000,tr,,) and it DOES exist in the output of the 'show dialplan' [ Context 'from-max' created by 'pbx_config' ] '_X' = 1. Answer() [pbx_config] 2. Queue(8000|tr||) [pbx_config] yet my system doesn't use it to route regards Tariq-- Maybe I missed a message or something, but I don't see a response to Tzafrir's request to see /etc/asterisk/extensions.conf. extensions_additional.conf is not extensions.conf; and unless extensions.conf includes it, it will never be a part of your dialplan. You did mention that you were using trixbox in your original question, so we referred you to a trixbox mailing list, because rumors have it that trixbox does complicated things in their dialplan to accomplish their goals, and most folks in this mailing list (but not all) don't play much with trixbox. But if you are not using trixbox, then you might look in your extensions.conf to answer these questions. Another resource you have to investigate the dialplan is in the CLI of asterisk; you can say dialplan show, or dialplan show from-max to see if the from-max context has been included. when the pbx_config module (module load pbx_config.so) loads, it will read in /etc/asterisk/extensions.conf; if it is not there, that module will not complete the loading process. If want us to evaluate why your dialplan is not working, show us the dialplan in extensions.conf. murf __ Date: Sun, 28 Sep 2008 23:31:46 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Issues On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/DAHDI ztdummy only
Tzafrir Cohen wrote: On Wed, Sep 24, 2008 at 03:23:52PM -0700, Roderick A. Anderson wrote: Let me know if I should post this on the asterisk-dev list instead. I am building a Linux-Vserver (http://www.linux-vserver.org) host system that will have several guests running Asterisk. Since the guests can't load kernel modules or do other dangerous stuff, but can access them I built zaptel 1.4 and it is now loaded by the host. modprobe ztdummy (alone) on the host. You'll have to create the basic device files on the gusts from the host. You'll have to use static device files. Sorry for taking so long getting back. By static device files you mean they are there when I enter the guest. I created them from the host system. (Ref: telephreak.org link below) Now I'm trying to build Asterisk in the guest but when I go through menuselect = Applications; app_flash and app_meetme are XXX'd out. They need Zaptel installed. A lsmod shows zaptel and friends are loaded. I'm assuming the guest really does have have access but the build processes look at something else to determine if Zaptel is installed. The Zaptel sources are installed and I did a: make libtonezone.so copied libtonezone.so to /usr/lib//usr/lib/libtonezone.so.1.0 and made a couple of links to give it more generic names; libtonezone.so.1 and libtonezone.so. This included putting a copy of zaptel.h in /usr/include/linux and tomezone.h in /usr/include. I found these instructions on the telephreaks.com site http://www.telephreak.org/papers/vpa/. Is there a way to over-ride make file build and force meetme and flash to be built? TIA, Rod -- The issue I see is there will be no Zaptel hardware and these guests will only do SIP, IAX, etc. but do appear to need ztdummy for timing with other services. So I'm looking for a way to not load (or even build) all the other modules that come as part of Zaptel. Possible? Yes. Will the complete change to DAHDI make this easier/harder? No. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
Ofcom banned end user revenue share on 070 numbers several years ago although the provider makes money. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: 29 September 2008 18:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] uk tole-free dids? On Mon, 29 Sep 2008, Babcock, Michael Alex wrote: what are 70 numbers? Prefix 070 (then 8 more digits) These are so-called personal numbers. They're a blot and an anomaly. They are expensive to call and the recipient usually gets revenue from the calls. ie. they are premium rate, revenue generating numbers in disguise. In disguise becasue a lot of people (in the UK) don't realise this because they look like mobile numbers - which start 07[1-9] then 8 more digits, so they think they're calling a mobile, when in-fact it's costing them much more. Gordon On Sep 28, 2008, at 10:26 PM, Geraint Lee wrote: You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
On Mon, Sep 29, 2008 at 11:15 AM, Steve Totaro [EMAIL PROTECTED] wrote: Huh, please try to form a complete thought I don't think I have to, since it's strictly an experience based questions, the answer I gave is what my experience has been. Grandstream can/will tell him that theirs works with some complete thought. And here is my thought, it just works, has all the features without a problem, and just don't need to be restarted. They will run and run and stay up unless you pull the power. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:53 AM, C F [EMAIL PROTECTED] wrote: Channel Banks only On Mon, Sep 29, 2008 at 10:44 AM, Steve Totaro [EMAIL PROTECTED] wrote: Quintum Tenor AX. Just glance over the manual. The are far better and in my experience just as reliable as a channel bank. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:26 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Oh, and not to mention, they are cheap. Very cheap in fact. On Mon, Sep 29, 2008 at 2:49 PM, C F [EMAIL PROTECTED] wrote: On Mon, Sep 29, 2008 at 11:15 AM, Steve Totaro [EMAIL PROTECTED] wrote: Huh, please try to form a complete thought I don't think I have to, since it's strictly an experience based questions, the answer I gave is what my experience has been. Grandstream can/will tell him that theirs works with some complete thought. And here is my thought, it just works, has all the features without a problem, and just don't need to be restarted. They will run and run and stay up unless you pull the power. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:53 AM, C F [EMAIL PROTECTED] wrote: Channel Banks only On Mon, Sep 29, 2008 at 10:44 AM, Steve Totaro [EMAIL PROTECTED] wrote: Quintum Tenor AX. Just glance over the manual. The are far better and in my experience just as reliable as a channel bank. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:26 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
bt give an annoying message before it connects your call, well, annoying if you actually are using 070 as a personal number and callers aren't charged stupid amounts of money to call it. virgin(old ntl) and h3g don't give any warning message at all though. 2008/9/29 asterisk [EMAIL PROTECTED] Ofcom banned end user revenue share on 070 numbers several years ago although the provider makes money. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: 29 September 2008 18:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] uk tole-free dids? On Mon, 29 Sep 2008, Babcock, Michael Alex wrote: what are 70 numbers? Prefix 070 (then 8 more digits) These are so-called personal numbers. They're a blot and an anomaly. They are expensive to call and the recipient usually gets revenue from the calls. ie. they are premium rate, revenue generating numbers in disguise. In disguise becasue a lot of people (in the UK) don't realise this because they look like mobile numbers - which start 07[1-9] then 8 more digits, so they think they're calling a mobile, when in-fact it's costing them much more. Gordon On Sep 28, 2008, at 10:26 PM, Geraint Lee wrote: You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
On Sep 29, 2008, at 9:55 AM, Yehavi Bourvine wrote: Try AudioCodes MP-124 which is 24 ports FXS. I have one but haven't used it much yet, so I cannot comment about its quiality. \ Sorry, cant agree with this, tried a couple and replaced with channel banks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
What is the best-recommended resource for searching archives of this mailing list? Thanks for your time ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel variables materializing ...
I am trying to track a strange bug down, and need to ask a really stupid
question, just so I can eliminate the possibility ..
When a SIP channel is hung up, I import a variable called MEETMEROOM
from the BRIDGEPEER channel, and if it is set, jump to another part of
the dialplan.
[snip]
exten = h,1,ImportVar(PARKED=${BRIDGEPEER},MEETMEROOM)
exten = h,n,GotoIf($[${PARKED} != ]?end)
exten = h,n,goto(DialStatus,${DIALSTATUS},1)
exten = h,n(end),NoOp()
[snip]
There have been several occasions over the past couple of days where
this variable has not executed the goto, and gone to the (end) label
when I know for certain that the BRIDGEPEER channel does not have the
variable set (I was able to duplicate the error once during a test phase
when I was not setting the MEETMEROOM variable at all)
so, to the stupid question: If at some stage the BRIDGEPEER channel
*has* had the MEETMEROOM variable declared, are there any circumstances
at all where this variable may be transmitted to the next call that uses
this channel.
There, I asked it. I don't believe that I just did. But there you have
it. It's out in the open now ...
The only other thing that I was thinking of - if the PARKED variable was
already set on the SIP channel, would an import of a non-existant
variable from the BRIDGEPEER channel overwrite it, or keep it at the
previous value ? Hmmm. Time to experiment.
Julian.
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Re: [asterisk-users] Source of SIP Remote host can't match request NOTIFY
I've been lucky enough to catch a NOTIFY/SUBSCRIBE sequence which ended with such 481 reply. (I don't have the slightest why NOTIFY is replied with 481 error, but that's another topic). Anyway, my question remains : Do you think message Remote host can't match request NOTIFY to call should be improved to Remote host 192.168.41.231 can't match request NOTIFY ... as, even with verbosity of 10, original message won't give any clue to find the responding host ? Regards 2008/9/29 Olivier [EMAIL PROTECTED] Hi, I've got a lot of Remote host can't match request NOTIFY. I've read http://lists.digium.com/pipermail/asterisk-users/2008-May/210709.html . Recommendation is enable SIP debug to see what packets are causing this, and if it's voicemail notifications, turn them off in sip.conf How would you proceed to get those debug ? The exact message is : WARNING[2990]: chan_sip.c:12900 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. As this 192.168.52.31 adresse is my Asterisk server address, I don't have a clue to focus debugging on the device that replies 481 to Asterisk. Unfortunately, I've got several dozens of devices and all with QUALIFY=yes. So in 60 seconds, I get hundred of SIP debug lines. Is there a smart way to handle this ? Something like SIP DEBUG 481 ? Anyway, do you think it could make sense to ask (or develop) a more explicit WARNING message such as Remote host 192.168.52.123 can't match ... ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel variables materializing ...
Julian Lyndon-Smith wrote:
I am trying to track a strange bug down, and need to ask a really stupid
question, just so I can eliminate the possibility ..
When a SIP channel is hung up, I import a variable called MEETMEROOM
from the BRIDGEPEER channel, and if it is set, jump to another part of
the dialplan.
[snip]
exten = h,1,ImportVar(PARKED=${BRIDGEPEER},MEETMEROOM)
exten = h,n,GotoIf($[${PARKED} != ]?end)
exten = h,n,goto(DialStatus,${DIALSTATUS},1)
exten = h,n(end),NoOp()
[snip]
There have been several occasions over the past couple of days where
this variable has not executed the goto, and gone to the (end) label
when I know for certain that the BRIDGEPEER channel does not have the
variable set (I was able to duplicate the error once during a test phase
when I was not setting the MEETMEROOM variable at all)
so, to the stupid question: If at some stage the BRIDGEPEER channel
*has* had the MEETMEROOM variable declared, are there any circumstances
at all where this variable may be transmitted to the next call that uses
this channel.
There, I asked it. I don't believe that I just did. But there you have
it. It's out in the open now ...
The only other thing that I was thinking of - if the PARKED variable was
already set on the SIP channel, would an import of a non-existant
variable from the BRIDGEPEER channel overwrite it, or keep it at the
previous value ? Hmmm. Time to experiment.
Julian.
__
This may be a long shot but would it not be better to check to see
whether or not the MEETMEROOM variable is defined before assigning it's
value to another variable? With just a cursory glance through the
asterisk documentation I have available I don't see any indication of
how asterisk variables behave if they are undefined.
The other possibility I was considering is maybe BRIDGEPEER is not
always being set to the correct channel?
Good luck,
-Brent
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[asterisk-users] Asterisk Documentation on voip-info.org (was: Re: Knowing incoming call technology and channel [SOLVED])
I've started the page at: http://www.voip-info.org/wiki-Asterisk+Documentation But I'm having problems with logging in via a script - I emailed [EMAIL PROTECTED] and J. Thompson has been very responsive in my request for help. I'll post back here when I've got something online to show. Cheers! -josiah Eric ManxPower Wieling wrote: You would want three pages, 1.2 docs, 1.4 docs, and 1.6 docs. Mark Hamilton wrote: I don't see why not, Voip-info is very outdated in most respects. Most of it with bad examples, dating to Asterisk 1.x era. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan Sent: September 29, 2008 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED] So, should we (I can do it, if desired) write a script that polls subversion docs directory and imports it into voip-info.org when the the docs are changed? I'd be glad to write and host such a script if the community desires the feature. -josiah SIP wrote: Eric ManxPower Wieling wrote: Olivier wrote: I don't have any spare zaptel enabled system I could try this on, but I was not aware of this CHANNEL variable. Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables Maybe, I will add a line in www.voip-info.org http://www.voip-info.org to keep others (me?) from searching again. You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt There's lots of cool information there, and all of it is up to date for your version of Asterisk, unlike voip-info.org. I often wonder why nobody seems to read the docs that are included with Asterisk. Web and/or context-searchable documentation will ALWAYS win out over a somewhat loose collection of text files. That's basic UI psychology 101. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Linksys SPA-8000 is an 8 port looks great, is there something similar which serves as a wireless router as well. On Mon, Sep 29, 2008 at 3:42 PM, Brian Webster [EMAIL PROTECTED]wrote: What is the best-recommended resource for searching archives of this mailing list? Thanks for your time ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] credit card processing
On Sat, Sep 27, 2008 at 6:52 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi Guys On the website, we already accept credit card by sending users to paypal website where we have an account. PayPal does have a service that is more like a traditional merchant service. I don't know if they have a real API that you can integrate into your system, however. Now, we want to do the same with an IVR where people can call a number, enter their credit card number and expiration date. This should be rather easy. Any traditional online merchant account. When you obtain a merchant account there are (simplified version follows) two parties involved, the bank that process the transactions and the gateway that accepts the transactions from the merchant (you) and sends them to the bank to be processed, in real time. Authorize.net is a very popular gateway supported by most e-commerce software. The point is that the Authorize.net API is a very popular system -- just about any pre-built e-commerce software supports it. It should be rather simple to create an AGI script which takes the credit card information and interfaces with the Authorize.net. They publish many examples and detailed API documentation so this should be a breeze for any skilled programmer. I strongly recommend that you use the CVV2 and AVS as a minimal means to reduce fraud. But I don't see any service or credit card procession company that offers this. What I want basicly is a service where I can send the credit card number I collected and expiration that and their charge the number and give me a status back. Do you know any company that do this ?? That's exactly the purpose of the Authnet API! Further information can be found here: http://developer.authorize.net/ Authorize.net also sells their gateway service under another name (I cant recall it right now), but everything else is the same. Also, some other gateways support Authorize.net emulation. Chris Bagnall wrote: Most credit card processing gateways require you to have the user's name and address for AVS verification when you perform customer not present transactions. Easy enough to do over a website, but a bit more tricky on the phone. AVS simply verifies the street number and zip code, nothing else. If I live at 123 Maple Street in zip code 77099 and I steal the credit card from someone at 123 Test Ct. in the same zip code I can have things mailed to me and it will pass AVS. Either way, when you are not shipping a physical product the rate of fraud rises dramatically -- you should carefully investigate fraud prevention for your system. Authorize.net provide a service which claims to flag/reduce fraudulent transactions. One of the merchant services I deal with, CDG Commerce (I highly recommend them, their customer service is top notch, but I dont think they will process for a VoIP/calling card service), has another similar system for no cost. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Thanks for the feedback. I'm particularly curious to know if anyone has tried a TDMoE channel bank. Spidermux seems to be one of the few vendors available. It's the closest I can get to an ATA-like device (ie. no special hardware, just ethernet) and it also offers an easy failover mechanism to another Asterisk server on the LAN. So I'm wondering why TDMoE channel banks aren't that popular (am I wrong?)? Is Asterisk's native TDMoE implemntation unreliable? Has anyone tested Spidermux or other TDMoE channel bank manufacturer? Standard channel banks become expensive when one has to buy the T1/E1 PCI cards on the Asterisk server. Since most channel banks interface with T1 (24 channels) and if I have about 350 analog phones to connect then I'd need around 15 T1s (that's about 4 quad-pri T1 cards which of course require 4 PCI slots and a fair amount of cash). Xorcom's Astribank is something in-between. It doesn't require PCI slots or T1 cards but connects via USB. Just like T1 channel banks, Astribanks don't seem to offer an easy failover mechanism like in the TDMoE solution (correct me if I'm wrong). However, a potentially higher number of astribanks can be cheaply connected to a single Asterisk server via several USB ports. I'm wondering how many 24-FXS astribanks can be connected via USB 2.0 to a single 4-USB-port server. Of the three solutions I'd try the TDMoE device but I'm wondering why noone in this thread even mentioned the protocol. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
I installed a few Spidermux units on some clients. It works fine, but I had some trouble with cordless analog phones connected to it. The ring voltage is pretty low and that caused some phones to not ring at all. And the ring voltage isn't configurable. Aside from that, it is a good product. Too bad it's the only TDMoE channel bank (that I know of, at least). Atenciosamente, Vinícius Fontes Núcleo de Tecnologias Convergentes Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Convergent Technologies Core Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Vieri [EMAIL PROTECTED] escreveu: Thanks for the feedback. I'm particularly curious to know if anyone has tried a TDMoE channel bank. Spidermux seems to be one of the few vendors available. It's the closest I can get to an ATA-like device (ie. no special hardware, just ethernet) and it also offers an easy failover mechanism to another Asterisk server on the LAN. So I'm wondering why TDMoE channel banks aren't that popular (am I wrong?)? Is Asterisk's native TDMoE implemntation unreliable? Has anyone tested Spidermux or other TDMoE channel bank manufacturer? Standard channel banks become expensive when one has to buy the T1/E1 PCI cards on the Asterisk server. Since most channel banks interface with T1 (24 channels) and if I have about 350 analog phones to connect then I'd need around 15 T1s (that's about 4 quad-pri T1 cards which of course require 4 PCI slots and a fair amount of cash). Xorcom's Astribank is something in-between. It doesn't require PCI slots or T1 cards but connects via USB. Just like T1 channel banks, Astribanks don't seem to offer an easy failover mechanism like in the TDMoE solution (correct me if I'm wrong). However, a potentially higher number of astribanks can be cheaply connected to a single Asterisk server via several USB ports. I'm wondering how many 24-FXS astribanks can be connected via USB 2.0 to a single 4-USB-port server. Of the three solutions I'd try the TDMoE device but I'm wondering why noone in this thread even mentioned the protocol. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maybe OT - routing calls in PSTN
I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine the responsible party and facilitating contact - because it is their DID/service that cannot be reached. In the past when I had a similar problem with a Junction DID, the folks at Junction resolved it with no hassles and zero intervention on my part. But Vitelity just keeps closing out my trouble tickets while responding in a way that indicates that they are not reading my reports carefully. How does this compare to others' experiences with Vitelity and other providers? Is there a way that I can determine whom to contact given only an originating number? Any words of wisdom? Documents I can read for educating myself? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe OT - routing calls in PSTN
If Vitelity is an ITSP, the problem is with the underlying carrier that provides the actual interconnection and switching facilities. It is their responsibility to contact the underlying origination carrier to resolve the issue. Bill Michaelson wrote: I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine the responsible party and facilitating contact - because it is their DID/service that cannot be reached. In the past when I had a similar problem with a Junction DID, the folks at Junction resolved it with no hassles and zero intervention on my part. But Vitelity just keeps closing out my trouble tickets while responding in a way that indicates that they are not reading my reports carefully. How does this compare to others' experiences with Vitelity and other providers? Is there a way that I can determine whom to contact given only an originating number? Any words of wisdom? Documents I can read for educating myself? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe OT - routing calls in PSTN
BTW, if you provide the originating number, the underlying carrier can be determined, either by the pooling or NANPA block it is assigned to, or its LRN if ported. If you want, you can privately e-mail me the number and I'll tell you who the carrier is. Alex Balashov wrote: If Vitelity is an ITSP, the problem is with the underlying carrier that provides the actual interconnection and switching facilities. It is their responsibility to contact the underlying origination carrier to resolve the issue. Bill Michaelson wrote: I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine the responsible party and facilitating contact - because it is their DID/service that cannot be reached. In the past when I had a similar problem with a Junction DID, the folks at Junction resolved it with no hassles and zero intervention on my part. But Vitelity just keeps closing out my trouble tickets while responding in a way that indicates that they are not reading my reports carefully. How does this compare to others' experiences with Vitelity and other providers? Is there a way that I can determine whom to contact given only an originating number? Any words of wisdom? Documents I can read for educating myself? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with my softphone
Hello, when with my client X-lite try to register in the server that say me, Registration error:501 Not implemented. What isn't implemented? the registration in the sip.conf or extensions.conf? how can i implemented that? thanks. Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: real 2 line phone vs. 1 line and call waiting
I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
