[asterisk-users] Unable to build smsq on beta6 and x86_64.
Hi, When I build the same asterisk package that I build on i386 on x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same set of installed packages. What should I be looking for in the output of ./configure to get a clue of what might be missing? TIA. -- Bill in Denver ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)
On Wed, 19 Mar 2008 10:10:21 + (GMT), Gordon Henderson [EMAIL PROTECTED] wrote: I got free installation for Featureline Compact on 3 yr contract. So it saved me £££s! Intersting... But shouldn't you be using VoIP for your calls anyway... Then just one basic BT line, and a business-quality ADSL service, then you can bypass all that nasty horrible analogy echoy stuff :) Problem is that we can only manage a measly 1.5 - 2Mb downstream here and we've already got ISDN2 for incoming. Outgoing are via my Trixbox which tries the sequence: ENUM SIP (we use Orbtalk) ISDN ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Thu, Mar 20, 2008 at 01:09:36AM -0400, Al Baker wrote: Not sure if this is the best place to ask this or not...but since it was mentioned.. Is SwitchVox a alternative to * ? Were they a competitor to *, and DIGIUM bought them and so DIGIUM has 2 Totally Different PBX software packages ? Think of Asterisk not as a PBX but as a PBX toolkit. Various people in this list build their PBX from this toolkit directly. Some of them do it for their home or company. Others resell it. Yet others use Asterisk as a part of a larger PBX. SwitchVox is one example of such a PBX. FreePBX is another. Druid is a third one. Digium also have hteir AsteriskNOW which is a very light-weight one. All of those packages are not an alternative to Asterisk (the PBX toolkit). They are an alternative to a home-grown PBX built on top of Asterisk. There are various pros and cons to both sides and various good and bad examples for both sides. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling 3 different call ending causes
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
On 3/17/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
You can handle the 'h' extension in the dial plan, which will supply
the
${CHANNEL} that was hung up, and possibly some additional dial plan
variables as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
Using these, you can piece together who hung up on whom, etc.
#2 is handled by fallthrough in the dial plan that causes the
instructions
to continue executing to the next priority for that extension, whereas
if the call completes (Dial() is successfully connected), this does not
happen.
I''ve tried to use the h extension in combination with the ${CHANNEL}
in the dialplan as suggested on the wiki page, but I haven't had any luck
with it.
For this test I have a Sipura phone with number 1003 and a X-lite with
1203.
If I let the time go by (the 20 seconds defined in the Dial Command) I
get the following:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
If I let the Sipura hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Lastly if I let the X-lite hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Yes they are all the same :(
Perhaps there's something wrong with my code? Its just a small
context with the following for this test:
[hangupcause]
exten = s,1,Dial(SIP/1203,30,gL(1[:5000][:5000]))
exten = s,2,NoOp(Callee hangup)
exten = h,1,NoOp(Channel hungup is ${CHANNEL})
Have I missed something basic here or what?
This should allow you to distinguish caller and callee hangups. I suppose
dial time limit will match Callee hangup, but you can check that by
${ANSWEREDTIME} or some sort of timestamp checking before and after Dial
(altough that would include ringing time)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
Hello List,
Ok, I solved it by using this code. This will work for me since the variable
${timeleft} is always in complete seconds. Thank you all for the ideas and
pointers :)
context hangupcause {
s = {
Set(timeleft=7000);
Dial(SIP/1203,30,gL(${timeleft}[:4000][:4000]));
if(${timeleft} = (${ANSWEREDTIME}*1000)) {
jump [EMAIL PROTECTED];
} else {
jump [EMAIL PROTECTED];
}
}
h = {
NoOp(Caller Hangup);
}
}
context hangupcause2 {
s = {
NoOp(Callee Hangup);
}
}
context notimeleft {
s = {
NoOp(Time's up!);
}
}
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Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Thu, Mar 20, 2008 at 12:59:08AM -0400, Alex Balashov wrote:
At the risk of inflaming a lot of passions, including those of
hard-working developers, I must say that where Asterisk may be
production-worthy, the entire constellation of things (like Zaptel) of
which its PSTN hardware interface capabilities comprise is absolutely
not, if my experience is at all telling. Of course, that's not all
Zaptel's or Digium's fault; much of it is just the buggy, flaky, and
very inconsistent nature of PC hardware, the kernel, ${insert true
culprit here}.
Nevertheless, my only truly solid experiences with Asterisk have come in
situations where it is used as a purely SIP agent. FXO interface
hardware, PRI cards (Sangoma, Digium, Rhino, etc.) all have bugs,
strange interop problems I've never seen before with big iron TDM
switches or newer telco softswitches that generate those circuits,
bizarre apparent interpretations of certain ISDN messages, and can cause
system instability, lockups, etc. (Whether they are the true cause of it
or whether that's just a consequence of their interoperation with the PC
is unknown to me, and somewhat beside the point.)
They've come a long way, I think. When I first used Digium T1 cards,
little, basic things like B channels not being hung up properly were
still a major and frequent theme. For low-capacity installs involving
at most one or two PRIs, I think one may be all right at this point.
But I still think it's experimental and avant garde from a production
standpoint; I find myself frequently stressing to my clients how much
better off they'd be just getting SIP termination and origination
elsewhere and breathe easier. Sangoma seems quite all right. Rhino is
OK, although as far as their multiport FXO interfaces go, it suffices to
say there is a difference between making it work and making it work
well. Their free support does go a long way toward that end. Your
mileage may vary. Caveat emptor.
Yeah, right. And we have no SIP compatibility issues at all. It is also
funny that you reflect the quality of old PRI card of one company and
yet ignore all the past mishaps of SIP devices.
I have stared long enough in both PRI traces and SIP traces. Both
protocols are complex. I've seen very strange things happening with SIP.
Also in this list. With Zaptel at least you have full ontrol of the
device
(disclaimer: I work for a Zaptel hardware vendor)
In general, though, almost any installation with any TDM trunking of
nontrivial volume is something in which I've ended up deploying
dedicated ISDN VoIP gateways, most recently Cisco AS5300s and 5400s. In
general, this is what I would advise to anyone thinking about
terminating more than a handful of PRIs, let alone DS3s worth of
traffic. Get proven, reliable hardware (even if it is expensive) from
vendors for whom TDM and carrier-grade telephony is a core competency.
I've seen far too many people try to take the cheap way out with a bunch
of Asterisk-oriented TDM hardware and not get quite what they were
expecting. Don't do it. There's something gravely perturbing about
running T1s into a PC anyway, although I know it's been done in certain
esoteric commercial telephony applications for eons.
Now please be specific about what is wrong with running a T1 into a PC.
I heard some people run Gigabit-ethernet into a standard PC. But maybe
that also takes a dedicated cisco gateway.
Pre.S.: while writing this I wanted to link to Jim Dixon's article The
History of the Zapata Telephony Project as it relates to the Asterisk
PBX. But it seems to have vanished off the face of the internet.
Anybody has a copy?
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Unable to build smsq on beta6 and x86_64.
On Thu, Mar 20, 2008 at 12:48:54AM -0600, William F. Acker WB2FLW +1-303-722-7209 wrote: Hi, When I build the same asterisk package that I build on i386 on x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same set of installed packages. What should I be looking for in the output of ./configure to get a clue of what might be missing? The thing we are missing are the errors you get. Prefferebly eith enough trace for us to have a clue. Cheers, -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Excellent topic and points brought up by all! On Thu, Mar 20, 2008 at 8:43 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Think of Asterisk not as a PBX but as a PBX toolkit. Various people in That's always been the way I saw asterisk. I wondered why people sometimes try to interface it with legacy pbx hardware, but over the years it became obvious that if you can get that working, it adds features to a reliable workhorse people are happy with. My small business has used asterisk built on hardware from the closet, a drive here, a mobo there, half a gig, two FXO cards and one TDM400P, 12 SIP or IAX providers, three phones and every SIP phone I can afford to mess with. It works very reliably until I try to do something to the dialplan I don't understand fully. Once I get that figured out and leave it alone, the box runs half a year before I reboot it on principle, or recently to replace the CPU fan. Bottom line, definitely ready for prime time for small operations IMO and a godsend! I'm currently playing with Digium's appliance and I hope to retire the old PC when we move in a few months. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Asterisk: Disaster Recovery Proof Asterisk
On Thu, Mar 20, 2008 at 01:27:47AM -0400, Al Baker wrote: From a lot of experience - you are not being anywhere near paranoid enough !! Think dual RAID controllers, Dual power supplies off of, at a Minimum, separate isolated circuits, with Hefty UPS that is in-line so it filters the power, think mirrored systems with a simple from switch from Black Box that will throw the lines over to the working box. Think -hmmm, how to keep v-mail between boxes in sync. Think remote console access to the server below multi-user level Think... is your azz is on the line for this install, I would give real consideration about doing this until you hzve a better plan, more $$$ in hand or a SLA to the end user that is very very forgiving. My 2 cents, worth all that you pad for it :) Hmmm. Isn't it simpler and cheaper to just keep a backup system? Dual- everything means a more complex system and hence more chances of something to go wrong (be that due to misconfiguration). Before jumping up and throwing hardware at the problem, think what the problem actually is. For instance, why a RAID? What are the important data we wish to keep? Can you afford loosing a couple of hours of records in case of such a disaster (from your last copy). Oh, and something in the news from yesterday: http://lwn.net/Articles/274079/ -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hint status unavailable
hello, i am trying to set up a asterisk server (version 1.2.26 by now) with realtime configuration but the user shouldnt register directly to the server, instead i have set up a ser registration proxy. Everything works fine so far, but i can´t use the hint feature. Its possible to subscribe to a given hint, but the status is allways unavailable and also i dont get a notify. Could someone help me finding a solution for this problem? I want to get notifies for hints where the user isnt registered on the asterisk itself. Thanks best regards Steve Smith ps: allready posted on Dev lists with the result this isnt a dev- related topic. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Line Status/Pickup
Hello, You have to set up a hint extension pointing to the Sip user like exten = 777,hint,SIP/username That extension is used in the Snom as extension. if you use the following format of this option field you should be able to pickup: sip:[EMAIL PROTECTED]|*9 777 is the hint extension 127.0.0.1 is your server ip |*9 is the extension to call for a pickup, so if you make a extension exten = *9777,1,Pickup(777) it should work best regards steve smith Brent Davidson schrieb: Does anyone know of a way to make a Snom 300 phone monitor the parking lot extensions and allow one-button pickup with the programmable buttons? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Für weitere Fragen stehen wir gerne unter [EMAIL PROTECTED] oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // [EMAIL PROTECTED] // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk re-invites and billing
I am using asterisk 1.4.18 (server A ) and have it store records in mysql database . One of my client uses predictive dialer ( asterisk 1.2.26 based and server B ) which makes many calls . B registers with A over sip and there is no nat involved If i re-invite rtp from server B to my carrier ( server A in between ) I saw many calls having duration of 0,1 or 2 seconds on server A's cdr but surprisingly all these calls were marked at 15 minutes usage on my provider's records . My sip route provider himself is re-inviting traffic ahead to their media gateways . I have gone through asterisk sip.conf and i don't see any setting limiting anything to around 15 minutes , default rtp timeout settings are around 60 seconds in asterisk . My provider says that they don't have any 15 minute limit on their end . The records on server B also suggests that calls are indeed very small 1 to- 3 seconds . Server A and B both have static ip's and there is no bandwith problem on server A . If i disable re invites on server A then this problem isn't present . Did anybody else have this kind of problem ? Any suggestions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to configure Voice mail for multi users.
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
thanks for you help,
regards,
Asif
Message: 14
Date: Wed, 19 Mar 2008 10:39:22 -0500
From: Eric Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] How to configure Voice mail for multi
users.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mian M Asif wrote:
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf . settings below..
[voicemail]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,n,NoOp(Dial Status: ${DIALSTATUS})
exten = _X.,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Background(vm-nobodyavail)
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
exten = s-NOANSWER,n,Hangup()
As I'm sure you know, ${EXTEN} is the value of the currently executing
extension, in the example above your line would be parsed as:
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED]) You would have
seen this if you were watching the Asterisk console when a call failed
to go to Voicemail.
Find some other way. You could save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1), but there are many, many,
many other ways.
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Re: [asterisk-users] Newbie IVR: How to read() before playback() is finished?
In article [EMAIL PROTECTED], Lee, John (Sydney) [EMAIL PROTECTED] wrote: I am working on a menu to accept input from a caller like as follows: Exten = 100,1,Answer() Exten = 100,n,Playback(LONG-MESSAGE) Exten = 100,n,Read(OPTION,,2) ... When I tested it, I noticed if I start pressing a key before the Playback() is finished, the input is not buffered (simply ignored) and I have to listen to the whole message before I could re-enter again. Is there a way that I could press a key and it will be Read() before the Playback is finished? Try this: exten = 100,1,Answer() exten = 100,n,Read(OPTION,LONG-MESSAGE,2) Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling 3 different call ending causes
On 3/20/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
On 3/17/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
You can handle the 'h' extension in the dial plan, which will supply
the
${CHANNEL} that was hung up, and possibly some additional dial plan
variables as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
Using these, you can piece together who hung up on whom, etc.
#2 is handled by fallthrough in the dial plan that causes the
instructions
to continue executing to the next priority for that extension, whereas
if the call completes (Dial() is successfully connected), this does not
happen.
I''ve tried to use the h extension in combination with the ${CHANNEL}
in the dialplan as suggested on the wiki page, but I haven't had any luck
with it.
For this test I have a Sipura phone with number 1003 and a X-lite with
1203.
If I let the time go by (the 20 seconds defined in the Dial Command) I
get the following:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
If I let the Sipura hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Lastly if I let the X-lite hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Yes they are all the same :(
Perhaps there's something wrong with my code? Its just a small
context with the following for this test:
[hangupcause]
exten = s,1,Dial(SIP/1203,30,gL(1[:5000][:5000]))
exten = s,2,NoOp(Callee hangup)
exten = h,1,NoOp(Channel hungup is ${CHANNEL})
Have I missed something basic here or what?
This should allow you to distinguish caller and callee hangups. I suppose
dial time limit will match Callee hangup, but you can check that by
${ANSWEREDTIME} or some sort of timestamp checking before and after Dial
(altough that would include ringing time)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
Hello List,
Ok, I solved it by using this code. This will work for me since the variable
${timeleft} is always in complete seconds. Thank you all for the ideas and
pointers :)
context hangupcause {
s = {
Set(timeleft=7000);
Dial(SIP/1203,30,gL(${timeleft}[:4000][:4000]));
if(${timeleft} = (${ANSWEREDTIME}*1000)) {
jump [EMAIL PROTECTED];
} else {
jump [EMAIL PROTECTED];
}
}
h = {
NoOp(Caller Hangup);
}
}
context hangupcause2 {
s = {
NoOp(Callee Hangup);
}
}
context notimeleft {
s = {
NoOp(Time's up!);
}
}
I would change that to = just for reliability - you never know :)
Regards,
Atis
--
Atis Lezdins,
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] Hardphone SIP phone costs
Gordon Henderson wrote: On Wed, 19 Mar 2008, Norman Franke wrote: As for why a company would purchase hard phones, several reasons. First, we are replacing many hard phones with computers. We have a custom application and have been moving folks main numbers to use the computer. We can make it ring externally and then they just put their headset on and hit an fkey to answer. The reason to not use a cell, in addition to potentially delaying an emergency response, is reliability. In any kind of emergency, they just don't work. And coverage and dropped calls are a problem, especially in office buildings. However, professionalism is, IMHO, the main reason. Cell phones sound terrible, generally have a huge delay (often with a related echo), they fade in and out, etc. I actively don't deal with companies where their sales people are on cell phones, and I have indeed actually to go with other vendors based on this. If you can't be professional enough to have an office with a real phone, why would I want to trust you''ll support anything you sell? My mobile does not sound terrible, does not have echo, does not fade in or out, and the last time I used it to call the emergency services, I got through straight away. I've not had a dropped call for a long time either (going through tunnels on the train, or over Dartmoor excepted) Sounds like your country doesn't have a very good mobile phone infrastructure, or operators that don't care. In the grand scheme of things, phone are cheap. With SIP phones, employees can move their phone to another office if they move and just plug it in. Companies can also better monitor employees. My mobile phone supports SIP (via WiFi) 3G and GSM... So I can move about and have coverage via a variety of means, but GSM just works in the UK, and when it doesn't work - well, I'm usually in a place where I don't want to make phone calls anyway :) It's not perfect and I'm still after the holy grail of the one device that will work anywhere, but it's getting there.. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Two of the four floors of the building in which I work are underground. Using cell phones as office phones has NEVER been a good idea. However, DECT phones just work. And well. We have cell repeaters installed inside the building, but the shielding in the walls limits their effectiveness in many areas (datacenter, conference rooms, etc). But again... no issues with the DECT phones. I don't think landlines will go away anytime soon simply for the very reason that mobile phones CAN'T be ubiquitous unless you have a tower or repeater anywhere where there's interference/low signal, etc. At our mountain cabin, we have a DSL line and VoIP is grand. But we'd have to drive six miles to get to the nearest spot with mobile coverage. I could never do any work from there if I relied on the mobile as my 'one true phone.' This isn't to say there's not a day when it will be better. But right now, it's patchwork at best. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Tzafrir Cohen wrote: Yeah, right. And we have no SIP compatibility issues at all. It is also funny that you reflect the quality of old PRI card of one company and yet ignore all the past mishaps of SIP devices. Oh, no, I didn't mean to imply that. There are plenty of SIP interop problems with Asterisk as well. I was actually just debugging a really recondite one yesterday with MetaSwitch. But nothing quite so dramatically abysmal as the TDM stuff. Of course, that could just be my particularly unfortunate experiences or shortcomings; I make no claim as to the universality of what I am saying. I have stared long enough in both PRI traces and SIP traces. Both protocols are complex. I've seen very strange things happening with SIP. Agreed, most certainly. In fact, it's funny how often I've heard that SIP is a simple protocol. Oh, you know, it's like HTTP, basically. Um, no, simple it is not. Now please be specific about what is wrong with running a T1 into a PC. I don't have a lot of specific objections, as I am not a hardware expert. I was just commenting on what seems to work well and what doesn't. If I had to speculate, there are backplane/bus throughput and timing differences between dedicated, embedded TDM hardware chassis with T1 interfaces and PC motherboards with offboard cards. One surely must be more imprecise, inconsistent and replete with compatibility problems than the other. I could be very wrong. I heard some people run Gigabit-ethernet into a standard PC. But maybe that also takes a dedicated cisco gateway. Ethernet is a data animal, not a synchronous voice animal. But then, a goat is not a synchronous voice animal either. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Wed, 19 Mar 2008 16:38:23 -0500, Bill Andersen [EMAIL PROTECTED] wrote: Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? /me raises hand. This said, if I did acquire sufficient knowledge of the system to be able to sell Asterisk-based solutions, I would probably do just that. -- Godwin Stewart - Horwich IT services ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? I'm responsible (development, maintenance, support) for an Asterisk based VoIP platform providing a replacement for residential PSTN lines. So I'm technically just a user ;-) I've literally got _thousands_ of users and Asterisk is rock solid for us. -- Andreas Sikkema ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Thu, Mar 20, 2008 at 06:45:14AM -0400, Alex Balashov wrote: Tzafrir Cohen wrote: Yeah, right. And we have no SIP compatibility issues at all. It is also funny that you reflect the quality of old PRI card of one company and yet ignore all the past mishaps of SIP devices. Oh, no, I didn't mean to imply that. There are plenty of SIP interop problems with Asterisk as well. I was actually just debugging a really recondite one yesterday with MetaSwitch. But nothing quite so dramatically abysmal as the TDM stuff. Of course, that could just be my particularly unfortunate experiences or shortcomings; I make no claim as to the universality of what I am saying. I have stared long enough in both PRI traces and SIP traces. Both protocols are complex. I've seen very strange things happening with SIP. Agreed, most certainly. In fact, it's funny how often I've heard that SIP is a simple protocol. Oh, you know, it's like HTTP, basically. Um, no, simple it is not. Now please be specific about what is wrong with running a T1 into a PC. I don't have a lot of specific objections, as I am not a hardware expert. I was just commenting on what seems to work well and what doesn't. If I had to speculate, there are backplane/bus throughput and timing differences between dedicated, embedded TDM hardware chassis with T1 interfaces and PC motherboards with offboard cards. One surely must be more imprecise, inconsistent and replete with compatibility problems than the other. I could be very wrong. PC hardware is produced in mass quantities. Hence you get hardware that is much more powerful. PCs today have hardware that has basically all the required CPU to handle quite some traffic. PCI (and even USB...) has been shown to be good enough to pass T1-s. Even with the unoptimized high interrupt rate of Zaptel. There's plenty of room for improvements in Zaptel. But people live with it right now because the CPUs we have are powerful enough. I heard some people run Gigabit-ethernet into a standard PC. But maybe that also takes a dedicated cisco gateway. Ethernet is a data animal, not a synchronous voice animal. But then, a goat is not a synchronous voice animal either. One main application is getting that synchronous voice over to voip. For that applicaiton we can easily afford adding a few delays. Now what happens when you actualyl want synchronous voice? Faxes? Modems? You could choose to of-load all of that to the dedicated gateway. But why? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Andreas Sikkema wrote: I've literally got _thousands_ of users and Asterisk is rock solid for us. I think most of the instabilities are from the use of queues and mixmonitor/chanspy. I don't use either and have no real issues. I still restart the Asterisk service once a week though, but these scripts have been in place since the pre-1.0 age. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status unavailable
On 3/20/08, Stefan Schmidt [EMAIL PROTECTED] wrote: hello, i am trying to set up a asterisk server (version 1.2.26 by now) with realtime configuration but the user shouldnt register directly to the server, instead i have set up a ser registration proxy. Everything works fine so far, but i can´t use the hint feature. Its possible to subscribe to a given hint, but the status is allways unavailable and also i dont get a notify. Could someone help me finding a solution for this problem? I want to get notifies for hints where the user isnt registered on the asterisk itself. Thanks best regards Steve Smith ps: allready posted on Dev lists with the result this isnt a dev- related topic. What did you mean by realtime config? Realtime SIP users, realtime dialplan? If it's just SIP users, you should have some success with rtcachefriends=yes in sip.conf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie IVR: How to read() before playback() isfinished?
- Original Message - From: Lee, John (Sydney) [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 19, 2008 11:48 PM Subject: [asterisk-users] Newbie IVR: How to read() before playback() isfinished? I am working on a menu to accept input from a caller like as follows: Exten = 100,1,Answer() Exten = 100,n,Playback(LONG-MESSAGE) Exten = 100,n,Read(OPTION,,2) ... When I tested it, I noticed if I start pressing a key before the Playback() is finished, the input is not buffered (simply ignored) and I have to listen to the whole message before I could re-enter again. Is there a way that I could press a key and it will be Read() before the Playback is finished? It seems like a lot of IVR system in the market can doing that and I am wondering if I have missed something in Asterisk. Any thoughts? Use Read( ) app to play your LONG-MESSAGE ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status unavailable
20 mar 2008 kl. 09.32 skrev Stefan Schmidt: hello, i am trying to set up a asterisk server (version 1.2.26 by now) with realtime configuration but the user shouldnt register directly to the server, instead i have set up a ser registration proxy. Everything works fine so far, but i can´t use the hint feature. Its possible to subscribe to a given hint, but the status is allways unavailable and also i dont get a notify. Could someone help me finding a solution for this problem? I want to get notifies for hints where the user isnt registered on the asterisk itself. That is something we all want, but it doesn't work now unless you add a third party software. I haven't seen anything that solves the issue, but have a few ideas. The question here is how should one asterisk be able to know anything about devices it doesn't control? It's a pbx, not an artificial intelligence software. There is work going on in the development group to make it possible to apply a message bus between Asterisk servers so that Asterisk servers can share call states. When that is up and running and tested, it will be part of a future Asterisk release. So the answer in short is not possible today, maybe tomorrow Regards, /olle Edvina AB * Asterisk training * http://edvina.net Asterisk SIP Masterclass * Orlando, FL, USA April 21-25 2008 OpenSER and Asterisk training - one week with the experts - register today! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Appologies for top-posting. This is the most interesting thread in a
long time. Alex, yours is the most well considered opinion I've seen in
a long while. I exactlt reflects my own, moerw limited experience.
Thank you for chiming in.
Two weeks ago on the VOIP Users Conference weekly call we had as our
guest Pika Technologies who are a Canadian company that make T-1/E-1
and FXO/FXS cards for Asterisk. One of their statements was the fact
that they don't rely on Zaptel for their driver. They have their own
driver which is unrelated to Zaptel.
Does anyone here have any experience with this? Is it markedly better,
or just more obscure and so harder to support?
Michael
On Thu, 20 Mar 2008 00:59:08 -0400, Alex Balashov wrote:
Very interesting thread!
My general sense, being both a person of heavy UNIX systems programming
and modest telco background, and as an Asterisk enthusiast, is that
Asterisk itself is quite production-worthy as such. Experience suggests
that what is controversial about it from a business standpoint, in terms
of total cost of ownership, support, and dependability, are many things
rather ancillary to it that contribute to the overall experience of an
Asterisk-based system as a product. Some of these pitfalls have already
been pointed out with regard to the shortcomings of consumer-grade PC
hardware, hard drives, power supplies, etc.
In other words, it seems to me that you can't just throw up an Asterisk
box as such and have it perform to your expectations. My experience
with the few Asterisk based IP PBX appliances that claim to be thusly
turn-key has been very poor, although, in their defense, it's been a
while and I'm sure those platforms have come a long way. But overall,
the domain of expertise required to make Asterisk work well in an
environment demanding of high availability is of a scope considerably
beyond Asterisk itself, and amounts to a fairly broad nexus of network
engineering, *nix systems administration, and so on. Most generalised
-- and, to some extent, highly specialised -- IT savvy is required, as
can be true with anything open-source and not packaged as part of some
immaculate, embedded black box culturally or technically.
Asterisk works well if deployed in a manner that brings quite an array
of skills to the table in a rather comprehensive way. In and of itself,
it assures little. This conclusion is supported by the differences in
my effort expended to support and (re)engineer third-party Asterisk
installations of varying quality and sophistication. And of course,
what I am saying here applies to most other things as well. It is
possible to set up Apache or MySQL or Linux itself naively, from the
heart, as well, as many do, or to do it in a nuanced, refined manner
that is attentive to the specificity of tight production requirements
and capitalises upon considerable expertise.
All Asterisk setups in which I have been involved have generally
involved a from-scratch custom compile of Asterisk, zaptel (if
necessary), and very frequently - especially if the latter is required -
a hand-compiled kernel as well. I do not use Trixbox, any Asterisk
administration front-ends, IP PBX appliances, and so on. I can't really
comment on their respective merits, but even if I could, I feel strongly
compelled to point out that this would be more of a referendum on
particular vendors or integrators who have packaged Asterisk a certain
way than about Asterisk in principle, which is something several people
have already said. If all of the nuances of a hand-maintained Asterisk
configuration are observed, I think it's a pretty solid product in any
event, but it does increase total cost of ownership for my clients as
they have to find someone like myself or other Asterisk consultants on
this list with the knowledge and experience to do that sort of thing.
It's the same sort of dilemma that arises between investing a lot of
faith in a stock CentOS or Fedora install by someone who kind of knows
a bit about Linux vs. hiring a really knowledgeable Linux sysadmin,
where the limitations of the distribution don't really matter because
they're going to know what to do with it on a highly detailed level.
The latter obviously gets vastly superior results, but costs a lot more
money and time.
At the risk of inflaming a lot of passions, including those of
hard-working developers, I must say that where Asterisk may be
production-worthy, the entire constellation of things (like Zaptel) of
which its PSTN hardware interface capabilities comprise is absolutely
not, if my experience is at all telling. Of course, that's not all
Zaptel's or Digium's fault; much of it is just the buggy, flaky, and
very inconsistent nature of PC hardware, the kernel, ${insert true
culprit here}.
Nevertheless, my only truly solid experiences with Asterisk have come in
situations where it is used as a purely SIP agent. FXO interface
hardware, PRI
[asterisk-users] question on app_conference()
MeetMe() has the K option that kills the conference, how do I do that in app_conference() as there no kill the conference option? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple sites, same extension
Must be having a DOH! week.
Problem turned out to be the Fedora core firewall that was turned on.
Sorry folks.
On Wed, Mar 19, 2008 at 3:01 PM, Aaron Fransen [EMAIL PROTECTED]
wrote:
Finally got my Cisco Call Manager link going; what it turned out to be was
having the same extension on the Asterisk system and on the Call Manager
side of things. Changing the extension on one side fixed it. Which brings me
to...
I need to have the same extensions on two sites. So if I use an 8bbb
dialing plan (8, then bbb location, then extension), site 1 might dial:
8-099-2000
Site 2 might dial: 8-101-2000
A 2000 extension exists on both sides, however Asterisk doesn't seem to
like it and drops the call before it even hits the logging facility and send
a busy signal back to the caller.
I've tried a dialing plan like:
exten = _8101,1,Dial(SIP/${EXTEN:4},,r)
to no avail.
Thoughts everyone?
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Re: [asterisk-users] hint status unavailable
On 20/03/2008, Johansson Olle E [EMAIL PROTECTED] wrote:
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything
works
fine so far, but i can´t use the hint feature. Its possible to
subscribe
to a given hint, but the status is allways unavailable and also i dont
get a notify.
Could someone help me finding a solution for this problem? I want to
get
notifies for hints where the user isnt registered on the asterisk
itself.
That is something we all want, but it doesn't work now unless you add
a third party software. I haven't seen anything that solves the issue,
but have
a few ideas.
The question here is how should one asterisk be able to know anything
about
devices it doesn't control? It's a pbx, not an artificial intelligence
software.
There is work going on in the development group to make it possible
to apply a message bus between Asterisk servers so that Asterisk
servers can share call states. When that is up and running and tested,
it will be part of a future Asterisk release.
So the answer in short is not possible today, maybe tomorrow
Regards,
/olle
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add it to the
end of the hint list to hint-the-hint. Something like:
exten = 234,hint,SIP/myphonePSEUDO/234
exten = *78,1,ChanAvailIs(PSEUDO/234,Busy)
exten = *791,ChanAvailIs(PSEUDO/234,Unknown)
This could be very useful for presence indication.
Cheers,
Steve
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Re: [asterisk-users] Is Asterisk ready for Prime-Time?
I reboot every evening :) Drew, what's the uptime on your asterisk process on that box that's been up for 193 days? I too restart the asterisk process every night as part of the cron process. Many people here seem to be under the impression that restarting the application every day is a bad thing. Having worked with carrier grade systems for 20+ years, I can tell you that even these systems restart the application during the days slow period. Granted these are usually two separate systems for redundancy but the typical method is to: 1) Unsync the two systems 2) Run system testing on the inactive side 3) Restart the inactive side 4) Resync the data between the systems 5) Switch the active and inactive processors 6) Repeat steps 1-4 on the newly inactive side Now don't think that smaller PBX or key systems are all that different. I know that the Meridian systems go through a similar process each day. On the SL1 systems there is a garbage daemon that runs every day. This daemon restarts the application to clean up RAM allocation. The Norstar key systems do this as well although the reset only takes about 2 seconds. Since everything is stored in flash memory, it is a quick way to make sure any glitches in RAM are cleaned up. Interestingly, the restart of Asterisk on my system only takes 3-4 seconds. Actual call processing is probably only affected for less that 2 seconds. Done during our night time activities no one ever notices. I've had some argue that a restart shouldn't be done because of the possibility that the system might not come back up. While this is potentially true, it will be because a file was changed without restarting or reloading asterisk. Yes this can happen though the likelihood is very small. At least it should be on a production system. One other thing to point out, if you are the type to constantly upgrade to the latest and greatest, you can expect to have issues. Once you get the system on a stable setup, the only reason for upgrading is if the new version fixes some problem that you have. Again some argue that security vulnerabilities would require the upgrade but that isn't always the case. If your system is a closed network, for example, your connection to the outside world is strictly analog and your network isn't shared with your computers, none of the security concerns would every matter. Now think back to that key system you revered for just working, did it have any outside connections that a hacker could exploit? Not likely. I have one system that we installed nearly a year ago. The only time it has been down was due construction workers cutting the main power feed to the building, between the building and the generator. It took them 10 hours to fix it and the UPS lasted over 4 hours. That was 200 days ago however asterisk was restarted about 8 minutes after midnight. As they say, your mileage may vary, but I don't think restarting asterisk is a bad thing. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status unavailable
On 3/20/08, Steve Davies [EMAIL PROTECTED] wrote:
On 20/03/2008, Johansson Olle E [EMAIL PROTECTED] wrote:
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything
works
fine so far, but i can´t use the hint feature. Its possible to
subscribe
to a given hint, but the status is allways unavailable and also i dont
get a notify.
Could someone help me finding a solution for this problem? I want to
get
notifies for hints where the user isnt registered on the asterisk
itself.
That is something we all want, but it doesn't work now unless you add
a third party software. I haven't seen anything that solves the issue,
but have
a few ideas.
The question here is how should one asterisk be able to know anything
about
devices it doesn't control? It's a pbx, not an artificial intelligence
software.
There is work going on in the development group to make it possible
to apply a message bus between Asterisk servers so that Asterisk
servers can share call states. When that is up and running and tested,
it will be part of a future Asterisk release.
So the answer in short is not possible today, maybe tomorrow
Regards,
/olle
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add it to the
end of the hint list to hint-the-hint. Something like:
exten = 234,hint,SIP/myphonePSEUDO/234
exten = *78,1,ChanAvailIs(PSEUDO/234,Busy)
exten = *791,ChanAvailIs(PSEUDO/234,Unknown)
This could be very useful for presence indication.
Huh, this hint hint would be useful for queues with local channel
state_interface too.. i think some general usage way could be added to
allow combining of device states.
Regards,
Atis
--
Atis Lezdins,
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] hint status unavailable
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add
it to the
end of the hint list to hint-the-hint. Something like:
exten = 234,hint,SIP/myphonePSEUDO/234
exten = *78,1,ChanAvailIs(PSEUDO/234,Busy)
exten = *791,ChanAvailIs(PSEUDO/234,Unknown)
This could be very useful for presence indication.
Huh, this hint hint would be useful for queues with local channel
state_interface too.. i think some general usage way could be added to
allow combining of device states.
Regards,
Atis
Machinations with func_devstate is the droid you're looking for.
However, there is an issue with the current use of state_interface in
app_queue where it is required to have a '/' character in it (obviously
would for Channels, but custom device states are of the form
Custom:yourdevicestate). I've worked around it, but I've been meaning
to file a bug report about it.
Anyway, have a look at that. It is being used successfully by us (in
1.4, with Russell's backported func_devstate and custom changes to fix
the aforementioned issue).
Regards,
- Brad
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Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? I actually work as a software engineer for a big telecom manufacturer to remain unnamed. I use Asterisk at home and I built a system for my aunt's real estate office mainly because she was quoted $83K+ over 5 years for a 12 station Toshiba key system. I now get calls to build more of them mainly for real estate offices that have seen other systems I have built. I probably should become a full blown reseller but I don't see me making enough money to walk away from my daytime gig anytime soon. On second thought, I guess if I were charging $83 large per customer maybe I could! John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Thu, Mar 20, 2008 at 09:31:03AM -0500, John Faubion wrote: I reboot every evening :) Drew, what's the uptime on your asterisk process on that box that's been up for 193 days? I too restart the asterisk process every night as part of the cron process. Many people here seem to be under the impression that restarting the application every day is a bad thing. It is a bad thing when people consider it a magic bullet. It cures slow resource leaks. But faster resource drains will still be able to crash Asterisk. Strange races will still happen once in a million. And people have actually suggested recently in this list a scheduled reboot as a cure for deadlock issues. And what happens if at the time of the shutdown there was a ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple sites, same extension
Holy Mackeral. Ignore that last message. I still do NOT know how to route
calls with the same extension being used in two locations, however the issue
I've resolved is getting Cisco CallManager and Asterisk talking together
properly.
Sorry folks AGAIN.
So if anybody has ideas on how to have extension 2000 in two places I'd
appreciate the advice! Thanks!
On Thu, Mar 20, 2008 at 7:14 AM, Aaron Fransen [EMAIL PROTECTED]
wrote:
Must be having a DOH! week.
Problem turned out to be the Fedora core firewall that was turned on.
Sorry folks.
On Wed, Mar 19, 2008 at 3:01 PM, Aaron Fransen [EMAIL PROTECTED]
wrote:
Finally got my Cisco Call Manager link going; what it turned out to be
was having the same extension on the Asterisk system and on the Call Manager
side of things. Changing the extension on one side fixed it. Which brings me
to...
I need to have the same extensions on two sites. So if I use an 8bbb
dialing plan (8, then bbb location, then extension), site 1 might dial:
8-099-2000
Site 2 might dial: 8-101-2000
A 2000 extension exists on both sides, however Asterisk doesn't seem to
like it and drops the call before it even hits the logging facility and send
a busy signal back to the caller.
I've tried a dialing plan like:
exten = _8101,1,Dial(SIP/${EXTEN:4},,r)
to no avail.
Thoughts everyone?
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[asterisk-users] Dialplan Help
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to check to see which they are actively
registered with, and send the call appropriately.
Right now I have:
Exten = _4xx,1,Dial(SIP/${EXTEN}IAX2/${EXTEN})
But not all phones have both techs, so there is a lot of misses
Is there a way to use the hints to see which they are registered with, and dial
only using those channel types?
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Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote: Sure some others on here may disagree, but I am also over on the trixbox forums, and have often seen talk about the 2.6.9 kernel having interrupt issues, and such that cause asterisk issues. One reason I think they moved forward into the CentOS 5.x stuff, so they got the 2.6.18 kernel, which I am told works much better, and doesn't have the issues the old kernel did. I've also found that I can't get ztdummy working on anything less than 2.6.23.11. Previous versions seem to have a broken RTC. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Mar 19, 2008, at 5:56 PM, [EMAIL PROTECTED] wrote: Anyone? Just a user? I'm just a user, although I also develop things for internal use. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to build smsq on beta6 and x86_64.
On Thu, 20 Mar 2008, Tzafrir Cohen wrote: On Thu, Mar 20, 2008 at 12:48:54AM -0600, William F. Acker WB2FLW +1-303-722-7209 wrote: Hi, When I build the same asterisk package that I build on i386 on x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same set of installed packages. What should I be looking for in the output of ./configure to get a clue of what might be missing? The thing we are missing are the errors you get. Prefferebly eith enough trace for us to have a clue. What's to trace, the file just wasn't built. Should I include the entire output of ./configure? Seems excessive to me, but if that's what you want... -- Bill in Denver ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] More DTMF issues
Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't seem to solve the problem. If I turn relaxDTMF off then most incoming calls from Cell Phones cannot navigate our menu and I get doubled digits showing up in the log. With it enabled 99% of calls work correctly, but a few of the calls do not show ANY dtmf digits being passed at all. I have debug and verbose both set to 20 and have several debugging NoOps set in the dial plan to add some call-flow tracking to the log. When someone calls in that has problems with DTMF our main greeting plays all the way through and the WaitExten times out and they are sent to our operator extension. I'm not 100% sure yet if this is relavant, but the majority of the people experiencing this problem are calling in from locations that have their own internal PBX systems. Is there maybe a way to further relax the DTMF detection? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
I was running Trixbox 2.2 up until about 2 months ago, and had persistent interrupt issues. I upgraded to 2.4, with the updated kernel, and its been complete smooth sailing ever since. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Norman Franke Sent: 20 March 2008 05:10 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Is Asterisk ready for Prime-Time? On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote: Sure some others on here may disagree, but I am also over on the trixbox forums, and have often seen talk about the 2.6.9 kernel having interrupt issues, and such that cause asterisk issues. One reason I think they moved forward into the CentOS 5.x stuff, so they got the 2.6.18 kernel, which I am told works much better, and doesn't have the issues the old kernel did. I've also found that I can't get ztdummy working on anything less than 2.6.23.11. Previous versions seem to have a broken RTC. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Thu, Mar 20, 2008 at 11:10:21AM -0400, Norman Franke wrote: On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote: Sure some others on here may disagree, but I am also over on the trixbox forums, and have often seen talk about the 2.6.9 kernel having interrupt issues, and such that cause asterisk issues. One reason I think they moved forward into the CentOS 5.x stuff, so they got the 2.6.18 kernel, which I am told works much better, and doesn't have the issues the old kernel did. I've also found that I can't get ztdummy working on anything less than 2.6.23.11. Previous versions seem to have a broken RTC. RTC? on kernel 2.6.23? It should not be used in kernel = 2.6.22 if CONFIG_HIGH_RES_TIMERS is enabled. Also note that on kernel 2.6.9 you have HZ=1000 and thus ztdummy works quite differently. This is not the same issue as interrupts of PCI devices. (not to mention that hotplug on CentOS 4 is strange. And earlier CentOS4 versions had a plain broken USB stack). -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] capacity
Thank you all for the great advice. Although fairly new to Asterisk, and relearning systems administration, it has helped put some perspective on the matter for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, March 19, 2008 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] capacity I use standard wav (most compatible with players) so about a meg a minute. In my experience, most people (users) use their voicemail similar to email, they keep everything. Especially love struck college kids. I think Asterisk has a soft limit of 1,000 (maybe it is 999) messages as the max per inbox that can be changed in source. I suppose if you limit the max time allowed and the max inbox limit it might help but I think your 60GB estimate would be quite low in the real world. BUT, that is based on when I was in college and I was one of the very few to have my own cell phone (dating myself a bit). So in the real world, I am not sure how much use the system would actually see. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 12:33 PM, Drew Gibson [EMAIL PROTECTED] wrote: Our office averages around 1.5MB / mailbox, call it 10MB for rounding. 6,000 x 10MB = 60GB (n'est pas?) 2 x 250GB drives, mirrored, should cover that and the system quite nicely. regards, Drew Disclaimer: Most of our employees are programmers so probably don't have any friends to call and leave messages! :-) Steve Totaro wrote: RAID arguments (preference really) aside, 4k - 6k worth of student voicemails is going to require quite a bit of storage space. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 12:01 PM, Drew Gibson [EMAIL PROTECTED] wrote: Having ventured high enough and far enough to view the curvature of the Earth and having stayed up late enough long enough (why do disks only fail at the weekend?) to rebuild and restore RAID 5 sets, I proffer the following (not so) Humble Opinion . Dual power supplies, two thumbs up but RAID 5 is only good for reducing storage costs on large volumes of data. It reduces performance and reliability over RAID 1. Don't put the OS on RAID 5 unless you like rebuilding servers from bare metal. It's much easier to rebuild and restore the data on RAID 5 sets if the OS is already up and running. Your OS and other system critical files (Asterisk) should be on RAID 1 for performance, redundancy and cost reasons. More disks = higher cost and higher chance of failure. Asterisk in general does not need much disk storage. The minimum drive size available in a new server tends to be overkill. Two drives as RAID 1 gives you redundancy and performance. Adding a third drive for RAID 5 adds cost, increases complexity and reduces reliability just to add storage capacity that you don't really need. (but the reseller WILL make more money and impress you with their command of the big words and acronyms on the spec sheet.) If and only if you need to store many hundreds of gigs of data (eg. recording a very large volume of calls) then RAID 5 becomes useful (or RAID 10 or RAID n). You should add this bulk storage IN ADDITION TO the mirrored pair holding the OS. regards, Drew Steve Totaro wrote: And I can post a link that shows a bunch of guys think the earth is flat with a 5/10 google ranking also (like the barf guys). http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm I usually just call my guy at CDW and give him my needs, he is a former techie gone sales. He puts together a quote and emails it to me for approval. I find HP server are very robust and rock solid at a decent price point (IBM as well). I like the 380 because you get six hot swap scsi bays and redundant power supplies in a 2u profile, also, Digium and Sangoma T1 cards have never given me an issue. Many on this list love Supermicro, I have yet to try them but I will in the near future. I have not heard a single complaint, only rave reviews. I guess my original point was going for redundancy as far as storage and power supplies with your dollar, not the fastest proc or maxed out RAM that will not be needed. Regardless of the actual hardware or RAID setup, that is the angle I suggest you take. 4k - 6k students will require quite a bit of storage. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 9:38 AM, Ron Joffe [EMAIL PROTECTED] wrote: On Tuesday 18 March 2008 22:12, Steve Totaro wrote: For your use, I would go for a RAID 5 I would highly recommend against a raid 5 set. I can give you more details if you are interested, but these guys have most if it down : www.baarf.com see
[asterisk-users] Polycom 650
List, Question about the Polycom 650: when dialing the digits for a phone number, and an incoming call comes in, does the phone prevent you from completing your outgoing call until the phone stops ringing, like a Cisco 79X0 does? --Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More DTMF issues
To add some further details to this thread I set up a Monitor command that records just the IVR portion of an incoming call. I left the m flag off so I could listen to the incoming audio separate from the outgoing recording. On calls where the DTMF detection works correctly I only hear extremely short clips of the DTMF tone in the incoming audio sample. They usually sound like just a click in the audio stream but there is a hint of the DTMF tone in the background. I'm assuming that Asterisk and/or my X100P is recognizing the DTMF tones immediately and muting them out of the audio path as soon as they are detected. What gets left behind is just the initial sample. The calls where DTMF detection is not working sound completely different. The tones in those audio streams are very loud and very distorted. It sounds almost like an old Atari 2600 video game instead of DTMF tones. The odd thing is that I can hear a hint of the correct tone in the background of the audio. When the caller is routed to the operator extension the voice audio quality is fine. Also, our outgoing greeting is noticeably quieter than the incoming DTMF tones so it shouldn't be interfering. I currently have rxgain set at 0.0 and have to set txgain to 4.5 or callers complain that our outgoing audio is too quiet. As much as I would love to say this is a problem with the Caller's PBX I just don't see how that's possible. They never had any problems calling us with our old phone system, and it is not limited to any one caller. Thanks, Brent Davidson Brent Davidson wrote: Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't seem to solve the problem. If I turn relaxDTMF off then most incoming calls from Cell Phones cannot navigate our menu and I get doubled digits showing up in the log. With it enabled 99% of calls work correctly, but a few of the calls do not show ANY dtmf digits being passed at all. I have debug and verbose both set to 20 and have several debugging NoOps set in the dial plan to add some call-flow tracking to the log. When someone calls in that has problems with DTMF our main greeting plays all the way through and the WaitExten times out and they are sent to our operator extension. I'm not 100% sure yet if this is relavant, but the majority of the people experiencing this problem are calling in from locations that have their own internal PBX systems. Is there maybe a way to further relax the DTMF detection? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
John Faubion wrote: Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? I actually work as a software engineer for a big telecom manufacturer to remain unnamed. I use Asterisk at home and I built a system for my aunt's real estate office mainly because she was quoted $83K+ over 5 years for a 12 station Toshiba key system. What on earth does that system do? open the office, make coffee, sweep the floors and ??? For such a small system there is no earthly reason for it to be 10 percent of that, even on a 5 year lease. Unless there are contract reasons she shouldn't even consider a lease either. I know that EVERYTHING is big in Texas, but that is nothing more than highway robbery. An NEC DSX with CF voicemail and e-mail integration wholesales for well under 3K, double that and add cabling . . . Well, you get the idea. John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple sites, same extension
Am Donnerstag, den 20.03.2008, 08:59 -0600 schrieb Aaron Fransen:
Holy Mackeral. Ignore that last message. I still do NOT know how to
route calls with the same extension being used in two locations,
however the issue I've resolved is getting Cisco CallManager and
Asterisk talking together properly.
I've tried a dialing plan like:
exten = _8101,1,Dial(SIP/${EXTEN:4},,r)
to no avail.
Hi Aaron,
for my personal taste your Dial() command is lacking a SIP domain (or IP
address). Consider location A (Asterisk 10.1.1.1, prefix 8101) and
location B (Asterisk 10.2.2.2, prefix 8202), where users at B want to
dial 81012000 for extension 2000 at location A.
In that case, your Dial command looks like
Dial(SIP/2000,,r), which looks pretty much useless, unless one of B's
local (see, local to B, not A!) SIP peers has a [2000] stanza in
sip.conf, and even then you would not call peer 2000 at A, but at local
(B).
If you replace your command with
Dial(SIP/${EXTEN:[EMAIL PROTECTED],,r)
the world looks completely different.
At least I hope so...
BR HTH
Anselm
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[asterisk-users] 423 Interval Too Brief and expiry settings in sip.conf
Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this error, but am not sure it really patching is necessary or can be avoided with different settings ? Thanks in advance, regards, Rob. --- (10 headers 0 lines) --- -- Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Mar 20 18:34:49] VERBOSE[7840] logger.c: --- SIP read from xxx.xxx.xxx.xxx:5060 --- SIP/2.0 423 Interval Too Brief Call-ID: [EMAIL PROTECTED] CSeq: 174 REGISTER From: sip:@ xxx.xxx.xxx.xxx;tag=as200dbc2c Min-Expires: 600 Server: Cirpack/v4.41f (gw_sip) To: sip:59972778@ xxx.xxx.xxx.xxx;tag=00-08013-1313fd60-3a4260273 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received= xxx.xxx.xxx.xxx;rport=5060;branch=z9hG4bK259bcd00 WWW-Authenticate: Digest realm=XXX.XXX,nonce=1313fbd315b1cefb52c870440e6f5455,opaque=1311d5ce56cc060,stale=false,algorithm=MD5 Content-Length: 0 - [Mar 20 18:34:49] VERBOSE[7840] logger.c: --- (10 headers 0 lines) --- [Mar 20 18:34:49] VERBOSE[7840] logger.c: -- Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx [Mar 20 18:34:49] VERBOSE[7840] logger.c: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF and Snom phones
Hello, I am having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to pickup the call. If I press the button next to the light the Snom phone doesnt do anything. It does not send a single packet to the network. I checked with sniffers and in the phones log. A manual dial to my pickup extension works of course. If the light is not blinking or not turned on and I press the button the phone dials the extension as it should. My phone setup: - latest snom firmware - function key set to Extension and content set to *7 which the phone converts to sip:[EMAIL PROTECTED];user=phone - on my asterisk *7 is configured and works Extensive googling and reading did not help. Any help is appreciated. I tested on different SNOM phones and the same Problem everywhere. Best regards, Loic Didelot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
For such a small system there is no earthly reason for it to be 10 percent of that, even on a 5 year lease. I know that EVERYTHING is big in Texas, but that is nothing more than highway robbery. I fully agreed, that's why we built her an Asterisk based system. Splitting this up they wanted $724 per month for the hardware and maintenance. This did include a special kind of lease where they could upgrade as necessary even if it required them to change out the system to do the upgrade. I'm not sure what that is worth but I'm fairly sure it shouldn't cost this much. The monthly contract for the Integrated PRI was another $675 per month. My aunt couldn't see how she was going to afford that so she called me for advice. I originally steered her toward a key system until I realized she would eventually need 35-40 stations. So we rolled our own asterisk based system. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
No, I meant if I leave this office, what to do when the cpu fan or power supply breaks on our current * box :) They might just be so worried that they'd *want* something like the 3Com V3000 :) Steve Totaro wrote: Call your dealer as I am sure you would have a support contract. Haven't really seen one break yet though. VxWorks is what runs satellites and junk ;-) Thanks, Steve Totaro On Wed, Mar 19, 2008 at 7:18 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Steve Totaro wrote: Anyways, as to the four FXO system, I would not think twice to steer that customer to the 3Com V3000. Interesting :) When I (the tech guy) leave this office, they just *could* be asking me what to do when it breaks? lol :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen: And what happens if at the time of the shutdown there was a ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ROTFL Trafrir, you made my day. (BTW: I think that is why restart when convenient exists) Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF and Snom phones
hello, you have to use following format in den extension key of the snom: sip:[EMAIL PROTECTED];user=phone|*7 the |*7 is the extension to dial if you want to pickup the ringing (blinking) line. maybe you should try sip:[EMAIL PROTECTED]|*7 where 100 is your hint extension and *7100 is a defined pickup extensions. best regards. Steve Smith Loic Didelot schrieb: Hello, I am having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to pickup the call. If I press the button next to the light the Snom phone doesnt do anything. It does not send a single packet to the network. I checked with sniffers and in the phones log. A manual dial to my pickup extension works of course. If the light is not blinking or not turned on and I press the button the phone dials the extension as it should. My phone setup: - latest snom firmware - function key set to Extension and content set to *7 which the phone converts to sip:[EMAIL PROTECTED];user=phone - on my asterisk *7 is configured and works Extensive googling and reading did not help. Any help is appreciated. I tested on different SNOM phones and the same Problem everywhere. Best regards, Loic Didelot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure Voice mail for multi users.
Mian M Asif wrote:
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
exten = s,n,Set(OLD_EXTEN=${EXTEN})
Then later, just use ${OLD_EXTEN}
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Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On March 20, 2008 02:33:52 pm Anselm Martin Hoffmeister wrote: Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen: And what happens if at the time of the shutdown there was a ROTFL Trafrir, you made my day. Oh god, I didn't realize that wasn't a typo until you wrote that... Very well done, Tzafir. Professionally executed. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want to know Frequency and lenght of Frame
[EMAIL PROTECTED] wrote: I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Aren't all the frames in asterisk 20ms long, no exceptions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account [EMAIL PROTECTED] wrote: Al Baker wrote: Quote This code is pre-Asterisk 1.0... It processes quite a few calls daily, I have about 1,800 DID numbers pointed at it, Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into that box, 1,800 DIDs pointing to it sems like one hell of a congestion problem and a Dialplan thicker than War and Peace I said DID numbers, they point to a PRI trunk group to a T400P, then the calls go IAX2 to other boxes for processing based on NPA/NXX. IE: exten=_906586,1,Dial,IAX2/un:[EMAIL PROTECTED]/[EMAIL PROTECTED] And if anything comes in for something not configured this catches it exten = _NX,1,Dial,sip/sipdebug/s If you figure standard telco usage patters, 92 channels @ 25:1 ratio, I have quite a bit of headroom. -Ron You don't run into choppy audio with IAX that way? I see that alot and the simple solution is to switch to SIP, almost always solves the problem right away. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want to know Frequency and lenght of Frame
Mojo with Horan Company, LLC wrote: [EMAIL PROTECTED] wrote: I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Aren't all the frames in asterisk 20ms long, no exceptions? Isn't ilbc the exception ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polarity in zapata.conf
hi: In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i added polarity reversal property but for fxo number 4 i didnt add polarity reversal property but it still giving me on cosole that fxo number 4 is polarized (because the line on fxo number 4 is not polarized). what i want to do is to not let polarity reversal take effect on fxo number 4. that what i have in my zapata.conf: answeronpolarityswitch=yes hanguponpolarityswitch=yes signalling=fxs_ks context=wassim channel = 1-3 ; signalling=fxs_ks context=wassim channel = 4 Thanks in advance; _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Steve Totaro wrote: On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account [EMAIL PROTECTED] wrote: Al Baker wrote: Quote This code is pre-Asterisk 1.0... It processes quite a few calls daily, I have about 1,800 DID numbers pointed at it, Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into that box, 1,800 DIDs pointing to it sems like one hell of a congestion problem and a Dialplan thicker than War and Peace I said DID numbers, they point to a PRI trunk group to a T400P, then the calls go IAX2 to other boxes for processing based on NPA/NXX. IE: exten=_906586,1,Dial,IAX2/un:[EMAIL PROTECTED]/[EMAIL PROTECTED] And if anything comes in for something not configured this catches it exten = _NX,1,Dial,sip/sipdebug/s If you figure standard telco usage patters, 92 channels @ 25:1 ratio, I have quite a bit of headroom. -Ron You don't run into choppy audio with IAX that way? I see that alot and the simple solution is to switch to SIP, almost always solves the problem right away. Not really, but both ends have zaptel hardware. I'm really surprised IAX2 connects and functions to these 1.4 and 1.6 beta servers from a Pre 1.0 machine. -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polarity in zapata.conf
On Thu, Mar 20, 2008 at 09:09:05PM +0200, wassim darwish wrote: hi: In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i added polarity reversal property but for fxo number 4 i didnt add polarity reversal property but it still giving me on cosole that fxo number 4 is polarized (because the line on fxo number 4 is not polarized). what i want to do is to not let polarity reversal take effect on fxo number 4. that what i have in my zapata.conf: answeronpolarityswitch=yes hanguponpolarityswitch=yes signalling=fxs_ks context=wassim channel = 1-3 ; signalling=fxs_ks context=wassim channel = 4 Just override it with a newer value: signalling=fxs_ks context=wassim answeronpolarityswitch=yes hanguponpolarityswitch=yes channel = 1-3 answeronpolarityswitch=no hanguponpolarityswitch=no channel = 4 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Al Baker wrote: Quote This code is pre-Asterisk 1.0... It processes quite a few calls daily, I have about 1,800 DID numbers pointed at it, Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into that box, 1,800 DIDs pointing to it sems like one hell of a congestion problem and a Dialplan thicker than War and Peace I said DID numbers, they point to a PRI trunk group to a T400P, then the calls go IAX2 to other boxes for processing based on NPA/NXX. IE: exten=_906586,1,Dial,IAX2/un:[EMAIL PROTECTED]/[EMAIL PROTECTED] And if anything comes in for something not configured this catches it exten = _NX,1,Dial,sip/sipdebug/s If you figure standard telco usage patters, 92 channels @ 25:1 ratio, I have quite a bit of headroom. -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF on Cisco 7970
Hello All, I've been trying to get BLF working with Asterisk 1.6-LatestBeta, and My Cisco 7970 (Latest SIP Firmware). Has anyone successfully completed this? I got the patch to merge in from http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones With a bit of hackery to that code, it successfully compiled chan_sip for me. So, With 1.6 we have TCP Enabled on Asterisk, Phone functions and registers fine. I think all that is left is the SEPMAC.xml configuration required for BLF on the softkeys. Can anyone else share some insights to this with me? Also, Along the same lines, does anyone have documentation on what the possible FeatureID's are for the line/line configuration in the SEPMAC.xml ? So Far I know 9 shows a little phone, and 22 shows the Speed Dial keypad. I don't want to try every possible number here just to find out though :) Thanks, Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Thu, 20 Mar 2008, Norman Franke wrote: On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote: Sure some others on here may disagree, but I am also over on the trixbox forums, and have often seen talk about the 2.6.9 kernel having interrupt issues, and such that cause asterisk issues. One reason I think they moved forward into the CentOS 5.x stuff, so they got the 2.6.18 kernel, which I am told works much better, and doesn't have the issues the old kernel did. I've also found that I can't get ztdummy working on anything less than 2.6.23.11. Previous versions seem to have a broken RTC. It works fine... # uname -a Linux dsx 2.6.18DSX1-CN #8 PREEMPT Fri May 18 16:13:30 BST 2007 i686 GNU/Linux # lsmod Module Size Used by zttranscode 6408 0 ztdummy 2632 0 zaptel182788 4 zttranscode,ztdummy # zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8184 sample intervals 99.902344% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8184 sample intervals 99.902344% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8184 sample intervals 99.902344% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8184 sample intervals 99.902344% --- Results after 9 passes --- Best: 100.00 -- Worst: 99.902344 -- Average: 99.956597 This is running on a 1GHz VIA C3 processor - no RTC, custom compiled kernel and zaptel compiled from source... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure Voice mail for multi users.
On Thursday 20 March 2008 05:06:29 am Mian M Asif wrote:
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
thanks for you help,
regards,
Asif
Message: 14
Date: Wed, 19 Mar 2008 10:39:22 -0500
From: Eric Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] How to configure Voice mail for multi
users.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mian M Asif wrote:
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf . settings below..
[voicemail]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,n,NoOp(Dial Status: ${DIALSTATUS})
exten = _X.,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Background(vm-nobodyavail)
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
exten = s-NOANSWER,n,Hangup()
As I'm sure you know, ${EXTEN} is the value of the currently executing
extension, in the example above your line would be parsed as:
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED]) You would have
seen this if you were watching the Asterisk console when a call failed
to go to Voicemail.
Find some other way. You could save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1), but there are many, many,
many other ways.
the variable setting i'm not helpful with, but how about:
[context]
exten = 2200,n,Dial(${DEVICE},20,kKotTwW)
exten = 2200,n,Goto(vm,${EXTEN},1)
[vm]
exten = _X.,1,Exec(${IF($[${DIALSTATUS}
= BUSY]?VoiceMail(${EXTEN},b):VoiceMail(${EXTEN},u))})
exten = _X.,n,Playback(vm-goodbye)
exten = _X.,n,Hangup()
the only part that gets repeated for each exten are the two lines in [context]
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
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Re: [asterisk-users] Is Asterisk ready for Prime-Time?
That probably includes 5 years of support but still expensive. John Faubion wrote: Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? I actually work as a software engineer for a big telecom manufacturer to remain unnamed. I use Asterisk at home and I built a system for my aunt's real estate office mainly because she was quoted $83K+ over 5 years for a 12 station Toshiba key system. What on earth does that system do? open the office, make coffee, sweep the floors and ??? For such a small system there is no earthly reason for it to be 10 percent of that, even on a 5 year lease. Unless there are contract reasons she shouldn't even consider a lease either. I know that EVERYTHING is big in Texas, but that is nothing more than highway robbery. An NEC DSX with CF voicemail and e-mail integration wholesales for well under 3K, double that and add cabling . . . Well, you get the idea. John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help
Hello, Do your verify, the codecs, of both clients, in your sip.conf? What codec do you use? Best Regards On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a beat sounds, and then nothing else. In the console, I can see: *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2001-081dd6e0, ) in new stack -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2001-081dd6e0, 2000) in new stack Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts 000160, len 000160) -- SIP/2001-081dd6e0 Playing 'vm-intro' (language 'en') Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts 000320, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts 000480, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037721, ts 000640, len 000160) Got RTP packet from192.168.1.102:8000 (type 00, seq 06, ts 1373137124, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037722, ts 000800, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037723, ts 000960, len 000160) Is it the prolem? First it sends to the public address of the the router, then it sends to the virtual IP. Is this the problem that causing my to hear just one beat sound and then no audio? Problem 2: The problem is isolated from Problem 1, cuz I run the SIP client on the same machine as the server, so there should not be network problem. I recorded some voice mails and they are stored as .wav files ok. When I tried to hear back the message, It does not work. Is there any configuration that I have to go through to have Asterisk to play .wav file? Thank you very much in advance for all your kind help. Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help
Pete, I have never done it but it would seem that running a SIP client on your SIP server may be problematic. Also, 58.251.75.228 is certainly not on your 192.168.x.x subnet. Is your machine dual homed? Thanks, Steve Totaro On Thu, Mar 20, 2008 at 9:34 PM, Carlos Rojas [EMAIL PROTECTED] wrote: Hello, Do your verify, the codecs, of both clients, in your sip.conf? What codec do you use? Best Regards On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a beat sounds, and then nothing else. In the console, I can see: *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2001-081dd6e0, ) in new stack -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2001-081dd6e0, 2000) in new stack Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts 000160, len 000160) -- SIP/2001-081dd6e0 Playing 'vm-intro' (language 'en') Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts 000320, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts 000480, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037721, ts 000640, len 000160) Got RTP packet from192.168.1.102:8000 (type 00, seq 06, ts 1373137124, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037722, ts 000800, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037723, ts 000960, len 000160) Is it the prolem? First it sends to the public address of the the router, then it sends to the virtual IP. Is this the problem that causing my to hear just one beat sound and then no audio? Problem 2: The problem is isolated from Problem 1, cuz I run the SIP client on the same machine as the server, so there should not be network problem. I recorded some voice mails and they are stored as .wav files ok. When I tried to hear back the message, It does not work. Is there any configuration that I have to go through to have Asterisk to play .wav file? Thank you very much in advance for all your kind help. Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help
Sorry, I am tired and missed the virtual IP part. I am not quite sure what that means or why you are sending traffic to the routeable IP. Are you using a FQDN with external DNS or the IP in your client? Thanks, Steve Totaro On Thu, Mar 20, 2008 at 10:42 PM, Steve Totaro [EMAIL PROTECTED] wrote: Pete, I have never done it but it would seem that running a SIP client on your SIP server may be problematic. Also, 58.251.75.228 is certainly not on your 192.168.x.x subnet. Is your machine dual homed? Thanks, Steve Totaro On Thu, Mar 20, 2008 at 9:34 PM, Carlos Rojas [EMAIL PROTECTED] wrote: Hello, Do your verify, the codecs, of both clients, in your sip.conf? What codec do you use? Best Regards On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a beat sounds, and then nothing else. In the console, I can see: *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2001-081dd6e0, ) in new stack -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2001-081dd6e0, 2000) in new stack Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts 000160, len 000160) -- SIP/2001-081dd6e0 Playing 'vm-intro' (language 'en') Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts 000320, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts 000480, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037721, ts 000640, len 000160) Got RTP packet from192.168.1.102:8000 (type 00, seq 06, ts 1373137124, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037722, ts 000800, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037723, ts 000960, len 000160) Is it the prolem? First it sends to the public address of the the router, then it sends to the virtual IP. Is this the problem that causing my to hear just one beat sound and then no audio? Problem 2: The problem is isolated from Problem 1, cuz I run the SIP client on the same machine as the server, so there should not be network problem. I recorded some voice mails and they are stored as .wav files ok. When I tried to hear back the message, It does not work. Is there any configuration that I have to go through to have Asterisk to play .wav file? Thank you very much in advance for all your kind help. Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
