RE: [Asterisk-Users] PLC (Packet loss cancel) questions
On 27 Aug 2004 at 5:56, Kevin Walsh wrote: [EMAIL PROTECTED] wrote: On 27 Aug 2004 at 2:33, Kevin Walsh wrote: There is no packet loss concealment in Asterisk at this time. Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? I'm note sure what you're referring to with the 1000 interrupts per second. Asterisk, as it stands, only reacts to incoming frames. If nothing is received then nothing is sent. The authors obviously didn't take packet loss into consideration. Ah, yeah the 1000 interrupts was referring to the interrupts generated off the digium cards (and why they don't much like interrupt sharing)... Yeah I know, it has implications in silence detection as well as PLC. I kinda meant why doesn't someone fire off whenever the interrupts add up to 20ms (or whatever size packet the voip traffic is coming in in) and process packets then...(not that I know much about the internals). When a packet is received, the expected time of the next packet is calculated. A while ago, I proposed that some sort of empty frame frame could be scheduled for now + next ETA. The arrival of the empty frame would wake up the receiver and, with the help of the jitter buffer, it could determine whether to pass on that frame to the translator, or to drop the packet as a duplicate. Some codecs could recognise the empty frame as a trigger to run their perform packet loss concealment code, whereas others (with no PLC) could simply treat it as a silent frame. Hmmm, this sound good. This all seems possible to me, but I haven't seen a discussion relating to this proposal nor any other alternatives. Take this over to -dev? Matt Riddell -- SNIPPED REST -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Echo SIP-T100P-PRI
On Aug 19, 2004, at 5:33 PM, Rich Adamson wrote: Mike Schwartz wrote: I'm experience echo on outgoing calls: Snom 200 Asterisk T100P PRI called party I am getting echo on the Snom 200 phone. The called party does not hear the echo. snip When that discussion was going on a few weeks ago, the echo issue seemed to have been narrowed down to two possibiliites; 1) interrupt service latency, or, 2) PCI bus latencies. Processor speed does not seem to be a driving factor as noted above. I've not heard anyone (as yet) come up with the tools or process for actually identifying the root-cause. Would be nice for those of us that aren't programmers. Some more echo food for thought. It's most noticeable on very short, hard sounds (like CH), so as someone mentioned, reverb might be the right description. I've spent the better part of several hours experimenting with various combinations of adjusting taps from 32 to 256, echowhenbridged on and off and txgain adjustments. I just flat can't get it to go away... I'm also one of those luck ones with a Supermicro box (dual Xeons and plenty of RAM). How in the heck would/should I go about figuring out what the interrupt service latency or the PCI bus latency is doing. Any other thoughts on the front? I'm using GS phones so maybe their echo can algorithms are to blame... hmmm... Here's to hoping, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: X-SBClass: Spam ( Re: [Asterisk-Users] TDM400P Problems)
I have a problem in that the pci card is not detected at all. Other pci cards are detected fine but not the tdm400b. It's detected on a different - smaller - machine no probs. Any ideas of the magic bios setting to get it going? - Original Message - From: Lyle Giese [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, August 27, 2004 12:41 PM Subject: X-SBClass: Spam ( Re: [Asterisk-Users] TDM400P Problems) Have you checked to see if there is a newer bios for the motherboard? - Original Message - From: Greg Hulands [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 9:31 PM Subject: Re: [Asterisk-Users] TDM400P Problems Even when the tdm400 is the only card in the computer, it still has this problem. I'm not sure how I would go about determining what is causing it. I have reset the bios to its factory settings to see if that helped, but alas it did not. Seems like i'm screwed. Greg On 27/08/2004, at 12:12 PM, Lyle Giese wrote: My guess is that you have PCI bus compatibility problems of some sort. Moving the cards around may help. Using a plug in NIC may help. A different Motherboard may help. This looks like hardware and trial and error and experience is all you have to lead you forward, if my guess is right. Lyle - Original Message - From: Greg Hulands [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 8:11 PM Subject: Re: [Asterisk-Users] TDM400P Problems When I did the lspci -v when it got to the tdm400p it seemed to go into an infinite loop on this line: Capabilities: [80] #00 []. When I redirected output to a file it filled to 8MB in about 3 seconds with this line. I moved the NIC so that it would be before the tdm400p, but it still did the same thing. I haven't a clue what is going on here. Any help is greatly appreciated. Regards, Greg Here is the output: 00:00.0 Host bridge: nVidia Corporation nForce2 AGP (different version?) (rev c1) Flags: bus master, 66Mhz, fast devsel, latency 0 Memory at d000 (32-bit, prefetchable) Capabilities: [40] AGP version 2.0 Capabilities: [60] #08 [2001] 00:00.1 RAM memory: nVidia Corporation nForce2 Memory Controller 1 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.2 RAM memory: nVidia Corporation nForce2 Memory Controller 4 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.3 RAM memory: nVidia Corporation nForce2 Memory Controller 3 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.4 RAM memory: nVidia Corporation nForce2 Memory Controller 2 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.5 RAM memory: nVidia Corporation nForce2 Memory Controller 5 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:01.0 ISA bridge: nVidia Corporation nForce2 ISA Bridge (rev a4) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0 Capabilities: [48] #08 [01e1] 00:01.1 SMBus: nVidia Corporation nForce2 SMBus (MCP) (rev a2) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel, IRQ 12 I/O ports at bc00 Capabilities: [44] Power Management version 2 00:02.0 USB Controller: nVidia Corporation nForce2 USB Controller (rev a4) (prog-if 10 [OHCI]) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 11 Memory at e4002000 (32-bit, non-prefetchable) Capabilities: [44] Power Management version 2 00:02.1 USB Controller: nVidia Corporation nForce2 USB Controller (rev a4) (prog-if 10 [OHCI]) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 9 Memory at e4003000 (32-bit, non-prefetchable) Capabilities: [44] Power Management version 2 00:02.2 USB Controller: nVidia Corporation nForce2 USB Controller (rev a4) (prog-if 20 [EHCI]) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 5 Memory at e4004000 (32-bit, non-prefetchable) Capabilities: [44] #0a [2080] Capabilities: [80] Power Management version 2 00:06.0 Multimedia audio controller: nVidia Corporation nForce2 AC97 Audio Controler (MCP) (rev a1) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 12 I/O ports at c000 I/O ports at b000 [size=128] Memory at e400 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 00:08.0 PCI bridge: nVidia Corporation nForce2 External PCI Bridge
[Asterisk-Users] 1stwave
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Re: [Asterisk-Users] system reboot often?
Leif Madsen wrote: Would you mind maybe expanding upon the hardware configuration you are using and why? I, and I'm sure others, are curious as to what you are using. I haven't had to roll out any systems yet that require multiple Digium cards, but I'm sure the information would be quite useful as I've seen few posts regarding this issue. Sure. They are nothing special - Asus P4B533 motherboard, P4 2.4, 256MB RAM, 40GB Western Digital SE PATA and 3c905c network card. As to why, the CPU was best value at the time and 533 FSB had just been introduced. The other components all seem to be solid performers in their classes. I will soon add a second drive - software RAID1. Personally I would not be deploying these for commercial use, as I feel the uptime is still not there. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk WITH Swyx... Any Idea?
We use asterisk coupled with swyxIt via a gnugk (version 2.0.8) and oh323 module: both swyx server and asterisk register on the gnugk. asterisk receives sip calls from the exterior and routes them to the gk. I've set up a prefix on swyx so that if I prepend +996 to my phone numerb, the call gests routed to asterisk (which, in turn, strips the prefix and sends the call via iptel or iaxtel. H.323 phones register on swyx. SIP phones register directly to asterisk) -- Message: 11 Date: Wed, 25 Aug 2004 19:35:16 +0200 (CEST) From: Zineddin Karzazi [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk WITH Swyx... Any Idea? To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 --- Loek Gijben [EMAIL PROTECTED] schrieb: Hello Zineddin, Hi, I'm a student and my thesis work consist in testing Asterisk with Swyx(SwyxWare). Until 2 years I was a student too so I think I can still relate to your state of mind :^) My approach is to declare asterisk as h323 gateway for the Swyxserver using oh323 Plugin. Is there any possibility to connect Asterisk with Swyx? how? Commercially we took a look at Swyx, it has great Windows (Active Directory) integration. But despite Swyx told us for nearly a year now that SIP support was coming we still haven't seen anything yet. So we left Swyxware where it belongs: on the shelf ;-)) 95% of innovations in VoIP are based on SIP I wonder why uou want to set up a system like this. Merely for testing purposes? Or does it have real life implications? Yes!! It s just for testing Purpose. If calling with swyxit is imminent then you can bypass the Swyxserver alltogether. And the Swyxserver can be hung on PSTN also, so if you need the AD integration than you can bypass the * server. I?m Tryin to connect the PBX without using ISDN or any Hardware. i already have a Nikotel Account to Be reachable under a regular phone number. but this is not possible to achieve with Swyx because it is based on H.323 and not SIP. I know this does not answer you question, and I'm not into H323. But like all other * users I've plowed myself through the Wiki and Googled a lot for answers. So if even I stumble on H323 on Asterisk info then it must be possible for you too. IMHO it must be possible to set up a system you describe, my hunch would be installing * on a testPC with H323 support, then first try to attach some softphones (like Swyxit) and the route through Swyxserver. Already Done for testing purpose. * with OH323 Plugin, works with 12 PC Clients (using Xlite(SIP),OpenPhone(H.323),IAXCOMM)and can also make calls to PSTN and be reachable from outside the LAN. The Problem is,that i cant use the Swyxit/Handsets or the Swyx IP-Phones. Succes with graduation! Loek Gijben Remotica Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advice on BT ISDN Services (UK)
Linus Surguy wrote: This isnt actually at all correct, we certainly have a Business Highway line in PTP mode with MSN! (Although you are right in that this is the default) Ah, BT, don't you just love them. Speak to three people and get four answers. I have just spoken to the ISDN support team and this is what they say today: Home Highway - Only available in PTMP with no MSNs Business Highway - Only available in PTMP with up to 8 MSNs ISDN2e - Available in PTMP with up to 8 MSNs or PTP with as many DDIs as you want The engineer I spoke to swore blind that you cannot have a line in PTP mode *and* have MSNs on it - they are completely incompatible. Like I said, you've got to love them! Ho hum. Nick Barnes Senior IT Consultant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g729 codec
[EMAIL PROTECTED] wrote: Aug 26 16:53:16 NOTICE[-239408208]: frame.c:120 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end I'm also seeing this message streaming in the CLI. I think it might have something to do with endpoints misbehaving when they use VAD. Maybe sending a start VAD frame continously, instead of doing nothing or restarting normal media. -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?
Consider this dialplan fragment, where the call is being dialed into
[macro-process-routing] over an iax2 channel from another (same build)
Asterisk server:
[macro-process-routing]
; This is the entrypoint of the debug call but is also refered to by
Macro(process-routing) elsewhere in the dialplan
; XXX-NNN-6800
exten = _6800,1,Macro(6800-interceptor)
; This is matched when 8 is dialed during macro-6800-interceptor,s,4
exten = _8,1,Playback(welcome)
exten = _8,2,Hangup
[macro-6800-interceptor]
exten = s,1,DigitTimeout,2
exten = s,2,ResponseTimeout,7
exten = s,3,Answer
exten = s,4,Background(autoattendant-ivr/grtg-6)
; Play full after-hours greeting
exten = t,1,Goto(s,1)
exten = i,1,Goto(s,1)
; However, this is never be matched if 8 is dialed during (s,4)
above
exten = _8,1,Playback(typhoon)
exten = _8,2,Hangup
So, why does the DTMF detect jump out to [macro-process-routing] instead of
staying within [macro-6800-interceptor]? The output of '-vvv' during the
event (with macro-process-routing,_8 removed) goes something like:
-- Accepting AUTHENTICATED call from 10.0.40.140, requested format = 2,
actual format = 2
-- Executing Answer(IAX2/[EMAIL PROTECTED]/1, ) in new
stack
-- Executing Macro(IAX2/[EMAIL PROTECTED]/1,
6800-interceptor) in new stack
-- Executing DigitTimeout(IAX2/[EMAIL PROTECTED]/1, 2)
in new stack
-- Set Digit Timeout to 2
-- Executing ResponseTimeout(IAX2/[EMAIL PROTECTED]/1,
7) in new stack
-- Set Response Timeout to 7
-- Executing Answer(IAX2/[EMAIL PROTECTED]/1, ) in new
stack
-- Executing BackGround(IAX2/[EMAIL PROTECTED]/1,
autoattendant-ivr/grtg-6) in new stack
Aug 27 01:13:06 DEBUG[68621]: channel.c:1101 ast_settimeout: Scheduling
timer at 160 sample intervals
-- Playing 'autoattendant-ivr/grtg-6' (language 'en')
Aug 27 01:13:06 DEBUG[7175]: chan_iax2.c:5156 socket_read: Ooh, voice format
changed to 2
Aug 27 01:13:09 DEBUG[68621]: channel.c:1101 ast_settimeout: Scheduling
timer at 0 sample intervals
Aug 27 01:13:09 DEBUG[68621]: app_macro.c:145 macro_exec: Oooh, got
something to jump out with ('8')!
Aug 27 01:13:09 DEBUG[68621]: pbx.c:1811 ast_pbx_run: Oooh, got something to
jump out with ('8')!
Aug 27 01:13:10 WARNING[68621]: pbx.c:1913 ast_pbx_run: Invalid extension
'8', but no rule 'i' in context 'macro-process-routing'
Aug 27 01:13:10 DEBUG[68621]: chan_iax2.c:2335 iax2_hangup: We're hanging up
IAX2/[EMAIL PROTECTED]/1 now...
-- Hungup 'IAX2/[EMAIL PROTECTED]/1'
Bug, feature or other suggestions? Build is 'Asterisk
CVS-HEAD-08/13/04-10:37:13'
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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Re: [Asterisk-Users] Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?
Kris Boutilier wrote: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten = _6800,1,Macro(6800-interceptor) ; This is matched when 8 is dialed during macro-6800-interceptor,s,4 exten = _8,1,Playback(welcome) exten = _8,2,Hangup [macro-6800-interceptor] exten = s,1,DigitTimeout,2 exten = s,2,ResponseTimeout,7 exten = s,3,Answer exten = s,4,Background(autoattendant-ivr/grtg-6) ; Play full after-hours greeting if the 6800-interceptor is only referenced within the process-routing macro, and nowhere else in the dialplan, couldn't you just create a new context called [6800-interceptor] and change the process-routing macro to: exten = _6800,1,Goto(6800-interceptor,s,1) That might work. Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup() doesn't always when talking to Nortel Norstar over CT1 E M wink-start trunk line?
I've noticed a problem with calls to Hangup when talking to my Norstars over channelised T-1 EM trunk lines - it's been present since I started to fiddle with Asterisk last December and it's still present in 'Asterisk CVS-HEAD-08/13/04-10:37:13'. Specifically, when a call is connected to Asterisk from the Norstar DTI card to my T100p I get the following conditions depending on the dialplan: exten = _6869,1,Hangup ; T1 channel stays up on the Norstar, but silent. ; -vvv says: -- Hungup 'Zap/24-1' ; show channels says: 0 active channel(s) exten = _6869,1,Answer exten = _6869,2,Hangup ; T1 channel stays up on the Norstar, but silent. ; -vvv says: -- Hungup 'Zap/24-1' ; show channels says: 0 active channel(s) exten = _6869,1,Answer exten = _6869,2,Wait(1) exten = _6869,2,Hangup ; T1 channel correctly drops on the Norstar. ; -vvv says: -- Hungup 'Zap/24-1' ; show channels says: 0 active channel(s) Asterisk always thinks the calls completed gone. For what it's worth, I have my Discon timer in the Norstar T1 card programming turned right down to 60 from the default 460 with no change - though I'm not certain exactly what this does... I get this issue whenever a call traverses the dialplan and, for what ever reason, is hungup by Asterisk without having previously been answered for at least a fraction of a second. Perhaps because this is EM wink the handshake isn't completed until the Answer is executed, thus the Hangup leaves the Norstar stuck half way? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
On 26 Aug 2004, at 17:48, Nick Barnes wrote:
Benjamin asked:
I don't have a problem setting this up under Asterisk (that's
the fun part) but what I need is advice on what to ask for
from BT so I don't get the wrong lines / services and so that
it all works smoothly!
OK. You need one of the following:
Home Highway
Business Highway
ISDN2e
I can confirm that * works happily with all three - my office lines
are (for
various reasons, none of which apply any more!) on Business Highway.
Heh, good old BT. I've never tested voice over Business Highway, as
every BT engineer/support/sales person I've spoken to swore blind that
it wouldn't work - and in BT's eyes, if they say it won't work, it's
unsupported, therefore, if it breaks - you're on your own.
Also, I don't believe you can get the full range of 'BT Select
Services' or whatever they call them today on the Highway lines (things
like Call Deflection, and even caller id on the home highway lines, I
believe)
TBH, If the line is used for voice, and you don't want the other
analogue lines that come with highway, go for ISDN2e - it's a full ISDN
service, and you won't get moaned at when you use it for voice.
If you want sequential numbers, then you'll have to argue like mad to
get
them as MSNs (I managed to get a block of 5 sequential MSNs, but it
was hard
work!). DDIs are issued in blocks of 10 and are usually sequential.
DDI's are ALWAYS sequential - it's the main thing BT push about them -
and they'll also try and sell you DDI's based on that fact when you
want to order MSN's. Depening on the department/person you speak to,
you may/may not be able to get sequential MSN's, although BT state that
for seq. numbers you MUST have DDI's.
The other thing to watch out for when you order the lines is which
rental
option you take out. BT offer three different options ('start up',
'call
plan' and 'low start') each with different installation costs, rental
charges and call allowances - if you don't specify which one you want,
they'll pick one for you (and probably at random).
BE CAREFUL! The calling plans sound quite attractive, but irrespective
of what sales say - the call allowance ONLY covers local and national
calls, NOT non-geo, mobiles, or international calls. I've just had a
huge row with BT High Level Complaints (we deal with them so often now,
we just skip the standard complaints department and call them directly)
- the salespeople will tell you it covers every sort of call you make
on the line - they're talking out of their lower hole...
HTH,
Jon
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RE: [Asterisk-Users] Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?
Alas, the dialplan itself is more complicated than I quoted in the problem
case below: There is an outer context that gathers digits from DID trunks,
pads the patter to comply with the corporate dial plan and then goes into
[macro-process-routing], thus:
[infrom-did]
exten = s,1,DigitTimeout,2
exten = s,2,ResponseTimeout,5
exten = s,3,Answer
exten = s,4,Ringing
exten = _XXX,1,Macro(process-routing,6${EXTEN}) ; This turns the 3 digits
coming down the DID trunk into 4 digit dialplan numbers
Then, inside [macro-process-routing] there are a reasonably large (100)
number of different patterns which each handle custom routing strategies for
each DID number - they're seperated out from the [infrom-did] context
because [macro-process-routing] is itself a target for incoming calls from
other internal Asterisk servers, who're using 4 digit dialing.
Those DIDs with IVRs are intended to be spun off into seperate macros with
self contained logics (such as [macro-6800-interceptor]) so their digit
collection strategies don't collide. However, I'm limited to one IVR at the
moment because the digit collection entries have to be back up in the
[macro-process-routing] routing context.
So, no - unwrapping the submacros isn't really feasable.
:-)
-Original Message-
From: el Flynn [mailto:[EMAIL PROTECTED]
Sent: August 27, 2004 1:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digit detect during a Background() inside
a Macro wrongly jumps b ack to the calling context to match digits?
Kris Boutilier wrote:
[macro-process-routing]
{clip}
[macro-6800-interceptor]
{clip}
exten = s,4,Background(autoattendant-ivr/grtg-6)
if the 6800-interceptor is only referenced within the process-routing
macro, and nowhere else in the dialplan, couldn't you just create a new
context called [6800-interceptor] and change the process-routing macro to:
exten = _6800,1,Goto(6800-interceptor,s,1)
That might work.
Flynn
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[Asterisk-Users] Re: Asterisk mysql database
I have just completed a project where I have just implemented running
extensions.conf from mysql.
When the extension is called, the dialplan passes the extension together
with the calling number a to a perl agi script. The perl script does a
couple of sql queries and sets a few variables which tells asterisk how to
route the call.
Here is an example of the extensions.conf
exten = _X.,1,AGI(SetVariablesFromSipToZap.agi|${CALLERIDNUM})
exten = _X.,2,SetCIDName(${CallerIDFullName})
exten = _X.,3,SetGroup(SIP/${OriginalCallerIDNumber})
exten = _X.,4,GotoIf,$[${RecordCalls} = 1]?5:7
exten = _X.,5,AGI(SetRecordOutgoingFromSip.agi|${CALLERIDNUM},${EXTEN})
exten = _X.,6,Monitor(gsm,${MonitorFile},m)
exten = _X.,7,Dial(Zap/g1/${EXTEN})
exten = _X.,8,StopMonitor
exten = _X.,9,Hangup
This works pretty well for me.
Daniel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Kenney
Sent: 26 August 2004 23:29
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk mysql database
I would like to configure asterisk to run complete out of mysql everything
from extensions to voicemail has this been implemented yet and how kind of
problems have their been
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[Asterisk-Users] 'set verbose 3' or other way to get '-vvv' level debugging out of running background asterisk?
I'm having a dialplan problem on one host where trunks get pinned up flapping between 't' and 'i' states and start eating lots and lots of CPU (loadavg 4.00). I haven't been able to pin down the problem reading through extensions.conf and test calls haven't caught it yet either. Unfortunatly the offending trunks are FXO immediate start DID trunks so subsequent callers are falling into an essentially dead link. Not all trunks go out at once which makes me think it's got to be a user-generated fault. I'd like to be able to connect to the running asterisk, issue 'set verbose 3' as is documented in the wiki and get '-vvv' style real time state information - but 'set verbose' doesn't appear to do anything when I'm connected using 'asterisk -r'. Should it work the way I'm anticipating in 'CVS-HEAD-08/13/04'? Alternatly, how would I go about configuring things so the '-vvv' style state information is collected into a file from the backgrounded Asterisk? That way I could at least do a post-mortem on the dialplan logic after it next goes off to the races. Thanks. Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated app_mysql.c, enabling use of INSERT and UPDATE
Hi,
For those interested in using MySQL directly from extensions.conf, there's
already a source file floating around for using a MYSQL application to
do SELECT queries.
We're using the MYSQL app a lot in our exensions.conf, but we missed
support for queries that don't return a result like UPDATE or INSERT.
Here's an updated app_mysql.c which introduces the Execute command.
Sample:
exten = s,300,MYSQL(Connect connid ${HOST} ${USER} ${PASSWORD} ${DATABSE})
exten = s,301,MYSQL(Execute resultid ${connid} UPDATE table SET haveSetting = 1 WHERE
dnid=\'${CALLERIDNUM}\')
exten = s,302,MYSQL(Disconnect ${connid})
This somewhat mimics the way the Borland implements this type of queries
in their products like Delphi.
It is a quick hack, but we've been using it for a couple of days now and
have not seen any issues with it. (yet? ;-) )
Enjoy.
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11 3011 VH Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
app_mysql.c
Description: app_mysql.c
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Re: [Asterisk-Users] PLC (Packet loss cancel) questions
On Thu, 26 Aug 2004, Jorge Verastegui G wrote: Have the astesrisk and digium people implemented PLC? No Are they implmementing it now? I want to but just haven't got to it yet. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't flash 7960: P0S30200 .bin not found
When I try to flash my 7960 with SIP I get messages like this in the tftp server logfile: Aug 27 02:01:17 home tftpd[32590]: tftpd: trying to get file: P0S3-03-0-00 .bin and the phone says something similar on the display for a brief moment and puts a funny char where the space in the filename above is. Seems like around 1 in 4 of the 7960's I have flashed with SIP have this problem. Anyone know what is going on here? I have googled and checked the wiki many times and cannot find anyone with this problem but it has happened to me twice now. I have two unusable phones until I get this fixed. The other 6 flashed just fine. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgpSB5fBtYe52.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card
On Thu, 26 Aug 2004 09:25:55 -0600, Andrew Elchuk [EMAIL PROTECTED] wrote: Is a sound card needed in order to playback some of the asterisk sounds in /var/lib/asterisk/sounds when dialing out with an X100P? Thanks. No Sound card is requied ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO interfaces used in UK?
What FXO interface methods are folks using successfully in the UK? I'm looking for good, known-to-work solutions for commercial use for two PSTN trunks on an Asterisk box. Here's the options I have, as I see it: i) Two Digium X100Ps. Pro - cheap (c. £120), CE approved. Con - UK line impedance mismatch, with resulting echo problems, plus needs two PCI slots. ii) Digium TDM400P with two FXO modules. Pro - still fairly cheap (c. £200), and can be set to UK line impedance, plus only uses one PCI slot. Con - Not yet CE approved. iii)Voicetronix Openline 4. Pro - reasonable price (c. £310), CE approved, only uses 1 PCI slot. Con - UK line impedance mismatch. iv) FXO gateway: - Multitech MVP210. Pro - UK line impedance, no PCI slot needed, good local technical support, CE approved. Con - expensive (£590 list) - Mediatrix 1204. Pro - As above, plus not so expensive (£389 from Telappliant). Con - can't get to see manuals to check functionality until you buy it I'm leaning towards either the TDM400P with 2xFXO, or the Mediatrix 1204, though because of the cons I'd like to hear from folks that have successfully used these in this type of configuration in the UK before I shell out for them! -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Channel CLI
On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi [EMAIL PROTECTED] wrote: Also dialing out works like a charm, the only problem is that calling out asterisk is displayed on the called phone instead of the sip address of the asterisk box. In the general section of sip.conf use the following line fromdomain=sip.address.com Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can't flash 7960: P0S30200 .bin not found
look to your SIPDefault.cnf or SIPmacaddress.cnf on TFTP server if you have is correct file name in image_version: section PJ - Original Message - From: Tracy R Reed Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Friday, August 27, 2004 11:48 AM Subject: Can't flash 7960: P0S30200 .bin not found When I try to flash my 7960 with SIP I get messages like this in the tftp server logfile: Aug 27 02:01:17 home tftpd[32590]: tftpd: trying to get file: P0S3-03-0-00 .bin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Sip Channel CLI
Hello Jason, Friday, August 27, 2004, 12:18:23 PM, you wrote: JW On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi JW [EMAIL PROTECTED] wrote: Also dialing out works like a charm, the only problem is that calling out asterisk is displayed on the called phone instead of the sip address of the asterisk box. JW In the general section of sip.conf use the following line JW fromdomain=sip.address.com Tnx ! Do I also have to define a peer in sip.conf or the registration as S exten is sufficient? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
On Fri, 27 Aug 2004 09:50:37 +0100, Jon Fautley [EMAIL PROTECTED] wrote: Heh, good old BT. I've never tested voice over Business Highway, as every BT engineer/support/sales person I've spoken to swore blind that it wouldn't work - and in BT's eyes, if they say it won't work, it's unsupported, therefore, if it breaks - you're on your own. Also, I don't believe you can get the full range of 'BT Select Services' or whatever they call them today on the Highway lines (things like Call Deflection, and even caller id on the home highway lines, I believe) I use business Highway, (Home highway works but MSN's are not availiable and CLIP- Callerdisplay is not an option for the ISDN Line) I have 8 MSN's Callerdisplay, Plus 2 analogue numbers all works great through a fritz card. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO interfaces used in UK?
On Fri, 27 Aug 2004 11:15:07 +0100, David Gurr [EMAIL PROTECTED] wrote: What FXO interface methods are folks using successfully in the UK? Ditch FXO completely and use a BRI Solution much better quality. or use Digium TDM400P card with two FXO modules, and apply UK CallerID patches, In my opinion ISDN is the way to go. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
Well I've just called BT, the confirmed to me that MSNs can only work with PTMP and DDIs with PTP. As for the seqential MSN issue, they have assigned me the five I requested - but not sequential. They did explain to me that they were automatically assigned randomly. I asked them to escalate my request for sequential MSNs and they have, will get back to me by the end of the day (or so they say). Is there any reason (other than cost) to use MSNs over DDIs or the other way round? Thanks for your help everyone - has been very useful. Can anyone recommend a good book or online reference for ISDN? Cheers, Benjamin Johnson Nick Barnes wrote: If you want sequential numbers, then you'll have to argue like mad to get them as MSNs (I managed to get a block of 5 sequential MSNs, but it was hard work!). DDIs are issued in blocks of 10 and are usually sequential. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
Kevin Walsh wrote: [EMAIL PROTECTED] wrote: On 27 Aug 2004 at 2:33, Kevin Walsh wrote: There is no packet loss concealment in Asterisk at this time. Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? I'm note sure what you're referring to with the 1000 interrupts per second. Asterisk, as it stands, only reacts to incoming frames. If nothing is received then nothing is sent. The authors obviously didn't take packet loss into consideration. When a packet is received, the expected time of the next packet is calculated. A while ago, I proposed that some sort of empty frame frame could be scheduled for now + next ETA. The arrival of the empty frame would wake up the receiver and, with the help of the jitter buffer, it could determine whether to pass on that frame to the translator, or to drop the packet as a duplicate. Some codecs could recognise the empty frame as a trigger to run their perform packet loss concealment code, whereas others (with no PLC) could simply treat it as a silent frame. This approach also is not fully right. On a system that implements silence suppression and uses discontinuous transmission (DTX), the receiver has a very tough job. I know that the current implementation of Asterisk doesn't work well with silence suppression but this doesn't mean that the design of a solution shouldn't take into account the full scenario. Look at the RTP stack of the receiver. When a packet is received, there are two cases: a) An RTP packet carrying voice frames is received. In that case the decoder will play the voice frames. b) A CN (Comfort Noise) packet is received. In that case the decoder will generate background noise (or do nothing). Now the hard part. Nothing is received (while something was expected). These are the normal interpretations of this situation: a) The transmitter detected silence and sent nothing (Silence). The receiver knows it from the last packet received (a CN packet). b) The transmitter sent a packet but the packet was lost (Packet loss). The receiver knows it from the last packet received (an RTP packet). These conditions can be identified at the RTP stack and signalled to Asterisk through the use of a new frame type (as you propose above). But, of course these are not always correct and the following situations could also happen: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. These cases cannot be identified, so the receiver just can only guess about what really happened and act accordingly. This all seems possible to me, but I haven't seen a discussion relating to this proposal nor any other alternatives. I hope that the above issues will start a discussion and result to a solution, no just for PLC, but also for the DTX operation. [deleted] Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error Compiling MySQL Friends
On Friday 27 August 2004 00:57, imail wrote: same error :( I just cant seem to figure it out, it must be something very obvoius. Can someone please point me in the right direction? [...] elifeq ($(USE_SIP_MYSQL_FRIENDS),1) [...] Looking at the GNU Make manual, there does not seem to be a command elifeq nor elif, only else. B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PLC (Packet loss cancel) questions
Michael Manousos [EMAIL PROTECTED] wrote: Look at the RTP stack of the receiver. When a packet is received, there are two cases: a) An RTP packet carrying voice frames is received. In that case the decoder will play the voice frames. b) A CN (Comfort Noise) packet is received. In that case the decoder will generate background noise (or do nothing). Agreed. Now the hard part. Nothing is received (while something was expected). These are the normal interpretations of this situation: a) The transmitter detected silence and sent nothing (Silence). The receiver knows it from the last packet received (a CN packet). b) The transmitter sent a packet but the packet was lost (Packet loss). The receiver knows it from the last packet received (an RTP packet). Both of the above cases are identifiable using a line state flag. Asterisk can (a) continue to generate CN or (b) generate a new frame type to get the codec to handle the concealment - where possible. These conditions can be identified at the RTP stack and signalled to Asterisk through the use of a new frame type (as you propose above). But, of course these are not always correct and the following situations could also happen: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. I would propose that after x lost packets, Asterisk should treat all further lost packets as CN. The proceeding x packets should be interpreted as RTP packet loss and run through the concealment routine. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. In this case, there is nothing to conceal anyway, as the last received data was a CN packet. In this case, the CN state should be continued until an RTP packet is received and the line state can be changed. The difficult part to handle would be late or out-of-sequence RTP packets. These should be ironed out by the jitter buffer. Late, lost and juggled packets are to be expected when dealing with UDP. This all seems possible to me, but I haven't seen a discussion relating to this proposal nor any other alternatives. I hope that the above issues will start a discussion and result to a solution, no just for PLC, but also for the DTX operation. I hope so too. Asterisk would benefit greatly from these improvements. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P lockups (FXO)
On Thu, Aug 26, 2004 at 07:08:19PM -0600, Marty Mastera wrote: I thought I would repost this info, since it seems relevant to this thread and may have been missed before this thread started... Marty, I appreciate your repost of this. I had seen it already and have written to Digium. That was yesterday and I have not yet heard back from them. If I do not hear from them by this afternoon, I will call them and see what I can find out. Thanks! I too have had problems with the TDM400P (TDM04B as configured) - symptoms ranging from one of the ports not being answered (no indication of an incoming call on the CLI), to calls dropping and outbound callers dialing and getting dead air. Sometimes a unload/load of the modules was sufficient to get things working again, other times a full reboot was required. Ultimately it was determined to be a design bug in the either the TDM400P itself or the FXO modules plugged into it. Digium acknowledged that it was known issue and sent me a replacement card and modules. The card had been modified, evident from the jumper wire that been soldered between two points on the back of the card. I haven't had problems since installing the new card. I would recommend contacting Digium support and sending them the serial numbers from the card and modules... Here's my original post for reference: http://lists.digium.com/pipermail/asterisk-users/2004-August/060008.html -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
Kevin Walsh wrote: Michael Manousos [EMAIL PROTECTED] wrote: Look at the RTP stack of the receiver. When a packet is received, there are two cases: a) An RTP packet carrying voice frames is received. In that case the decoder will play the voice frames. b) A CN (Comfort Noise) packet is received. In that case the decoder will generate background noise (or do nothing). Agreed. Now the hard part. Nothing is received (while something was expected). These are the normal interpretations of this situation: a) The transmitter detected silence and sent nothing (Silence). The receiver knows it from the last packet received (a CN packet). b) The transmitter sent a packet but the packet was lost (Packet loss). The receiver knows it from the last packet received (an RTP packet). Both of the above cases are identifiable using a line state flag. Asterisk can (a) continue to generate CN or (b) generate a new frame type to get the codec to handle the concealment - where possible. These conditions can be identified at the RTP stack and signalled to Asterisk through the use of a new frame type (as you propose above). But, of course these are not always correct and the following situations could also happen: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. I would propose that after x lost packets, Asterisk should treat all further lost packets as CN. The proceeding x packets should be interpreted as RTP packet loss and run through the concealment routine. Well, no matter what kind of concealment algorithm is used, just the first one or two packets will be concealed. The rest losses will result in no-playback. No CN interpretation, just absolute silence. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. In this case, there is nothing to conceal anyway, as the last received data was a CN packet. In this case, the CN state should be continued until an RTP packet is received and the line state can be changed. Exactly. So the receiver, in case of no-receiption, should go back and see what was the last packet correctly received and act as I described above. The difficult part to handle would be late or out-of-sequence RTP Actually this is not so difficult, if there is a jitter buffer. packets. These should be ironed out by the jitter buffer. Late, lost and juggled packets are to be expected when dealing with UDP. This all seems possible to me, but I haven't seen a discussion relating to this proposal nor any other alternatives. I hope that the above issues will start a discussion and result to a solution, no just for PLC, but also for the DTX operation. I hope so too. Asterisk would benefit greatly from these improvements. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Touch tone problem
Group This is strange. When I call my voice mail extension the system does not pick up my touch tone entries. I have x-lite softphone and a cisco 7960 for my hard phone. When I call from outside I'm able to check my voice mail without any problem. Any help would be great! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
On Fri, 27 Aug 2004, Michael Manousos wrote: I hope that the above issues will start a discussion and result to a solution, no just for PLC, but also for the DTX operation. Yeah - my goal for a reworked jitter buffer includes DTX and PLC. And other TLAs ;-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PLC (Packet loss cancel) questions
Michael Manousos [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Michael Manousos [EMAIL PROTECTED] wrote: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. I would propose that after x lost packets, Asterisk should treat all further lost packets as CN. The proceeding x packets should be interpreted as RTP packet loss and run through the concealment routine. Well, no matter what kind of concealment algorithm is used, just the first one or two packets will be concealed. The rest losses will result in no-playback. No CN interpretation, just absolute silence. That's true - unless there's some logic to say that after x lost packets, the line state should switch to CN generation instead of silence. The line state would switch back once a fresh RTP packet is received. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. In this case, there is nothing to conceal anyway, as the last received data was a CN packet. In this case, the CN state should be continued until an RTP packet is received and the line state can be changed. Exactly. So the receiver, in case of no-receiption, should go back and see what was the last packet correctly received and act as I described above. Maintaining an audio state flag (CN/RTP) would be the key here. The difficult part to handle would be late or out-of-sequence RTP packets. These should be ironed out by the jitter buffer. Late, lost and juggled packets are to be expected when dealing with UDP. Actually this is not so difficult, if there is a jitter buffer. Right. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help to install ISDN Fritz card
Hi everybody, I need a litle help to install Asterisk using ISDN Fritz PCI card on my linux box fedora 1. All suggestions with links or samples are welcome. I would be really pleased for any help :) Radu, E-mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using regular expression in dialplan
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 = OK, Goto 4
else
Goto 2
I've tried the follwoing syntax but it is alway going to 2 whatever
the value of DTMFSeq:
exten = s,1,GotoIf($[${DTMFSeq} : 123]?4:2)
exten = s,2,SetVar(InvalidCount=$[${InvalidCount} + 1])
exten = s,4,SetVar(Result=ok)
The only way I managed to make it work is the following :
exten = s,1,GotoIf($[${DTMFSeq} : 1|2|3]?s|4:s|2)
But I'm not totaly satisfied with it as I'm going to check more
complex regex later ...
Thank you for your help
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Re: [Asterisk-Users] Asterisk to Vonage
When I initially signed up with Packet8 and they sent their converter, I used a X100P card in my Asterisk server so that it could send and receive calls through Packet8, I suspect the same trick would work for Vonage. The benefit is you can then have several phones in the house, or one at work, or a softphone in your laptop, register with your Asterisk server and then you can place and receive unlimited local/long distance calls through your Vonage account. You can also have Asterisk answer and you can use it's IVR/Automated Attendant functionalities. You will be limited to only 1 inbound/outbound call at a time though. I eventually canned Packet8 in favor of BroadVoice, $19.95 a month, unlimited local and long distance, and I've tested up to 6 inbound calls at the same time and it worked. Ask Bjørn Hansen wrote: On Aug 24, 2004, at 4:17 PM, Chris Shaw wrote: I hold no ill will towards Vonage but I have to say honestly... ewww... They've already made their feelings quite clear by refusing to allow people to bring their own devices and taking steps to even hide their SIP servers (changing the port from the RFC standard 5060 to 5061 for example.) Why not go with someone who's actually willing to allow you to use Asterisk and any phone you want like NuFone, BroadVoice, IconnectHere or a host of others instead of trying to hack Vonage... At least when I signed up with Vonage they were the only VoIP provider that had numbers in my old rate center and could transfer the number from SBC. It does, of course, suck not to be able to use it with Asterisk. (I could sign up for a soft-phone, but I don't think it'd be with my old number defeating the purpose...) - ask ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Echo SIP-T100P-PRI
Have you considered relocating the hard drive (or Asterisk configs) and the T100P card to a temporary machine? Even a lower class machine, just to eliminate the SuperMicro as the possibility? I'm interested in your research as we will be deploying some low end $800 1U (very short) SuperMicro servers out in the field equipped with T100P cards in them. They have Celeron 2ghz processors, 512mb of RAM and a regular IDE hard drive. But if there is an incompatibility with the way SuperMicro makes their motherboards, then I need to stop them from placing the order for 10 of these units on Monday. Ryan Thrash wrote: On Aug 19, 2004, at 5:33 PM, Rich Adamson wrote: Mike Schwartz wrote: I'm experience echo on outgoing calls: Snom 200 Asterisk T100P PRI called party I am getting echo on the Snom 200 phone. The called party does not hear the echo. snip When that discussion was going on a few weeks ago, the echo issue seemed to have been narrowed down to two possibiliites; 1) interrupt service latency, or, 2) PCI bus latencies. Processor speed does not seem to be a driving factor as noted above. I've not heard anyone (as yet) come up with the tools or process for actually identifying the root-cause. Would be nice for those of us that aren't programmers. Some more echo food for thought. It's most noticeable on very short, hard sounds (like CH), so as someone mentioned, reverb might be the right description. I've spent the better part of several hours experimenting with various combinations of adjusting taps from 32 to 256, echowhenbridged on and off and txgain adjustments. I just flat can't get it to go away... I'm also one of those luck ones with a Supermicro box (dual Xeons and plenty of RAM). How in the heck would/should I go about figuring out what the interrupt service latency or the PCI bus latency is doing. Any other thoughts on the front? I'm using GS phones so maybe their echo can algorithms are to blame... hmmm... Here's to hoping, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues - CallbackLoging Automaically?
Been trying to set up a call queue with agent call back without the need for the agent to have to log in. Have set up the queue sucessfully. However I want to remove the requirement for agents to have to log in as they are on static extensions. Is there a way of either using extentions in the queue.conf instead of agents? Or an automatic way of logging in the agent to the callbackloging function? Have come up on a dead end reading up and would appreicate any help.. Andrew Brown ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 - SCCP or SIP?
Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems this is where some sort of black magic needs to be used as I cannot find any way of getting them to work which leads me to the main question Is it better to use chan_sccp or SIP? I know these phones can work in either mode I was just wandering which is the better format and which has the most functions implemented? Its a simple home environment that I am planning but it would be good to be able to use the softkeys to transfer calls and to pickup messages. Thanks in advance, Sam Kevin Walsh [EMAIL PROTECTED] wrote on 27/08/2004 13:59:09: Michael Manousos [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Michael Manousos [EMAIL PROTECTED] wrote: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. I would propose that after x lost packets, Asterisk should treat all further lost packets as CN. The proceeding x packets should be interpreted as RTP packet loss and run through the concealment routine. Well, no matter what kind of concealment algorithm is used, just the first one or two packets will be concealed. The rest losses will result in no-playback. No CN interpretation, just absolute silence. That's true - unless there's some logic to say that after x lost packets, the line state should switch to CN generation instead of silence. The line state would switch back once a fresh RTP packet is received. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. In this case, there is nothing to conceal anyway, as the last received data was a CN packet. In this case, the CN state should be continued until an RTP packet is received and the line state can be changed. Exactly. So the receiver, in case of no-receiption, should go back and see what was the last packet correctly received and act as I described above. Maintaining an audio state flag (CN/RTP) would be the key here. The difficult part to handle would be late or out-of-sequence RTP packets. These should be ironed out by the jitter buffer. Late, lost and juggled packets are to be expected when dealing with UDP. Actually this is not so difficult, if there is a jitter buffer. Right. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ [EMAIL PROTECTED] _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Winckworth Sherwood Solicitors and Parliamentary Agents DX 148400 WESTMINSTER 5 : 35 Great Peter Street, London SW1P 3LR Telephone 020 7593 5000 Fax 020 7593 5099 Confidentiality This email message and any attachments are confidential; they may be subject to legal professional privilege and are intended for the named recipient only. If you are not the named recipient, please return the message and enclosures immediately and delete them from your system. Caution Before advice received only by email (whether by attachment or otherwise) may be relied on, the authenticity of the communication must be verified by means independent of email. Regulation The firm is regulated by the Law Society. Partners A list of partners is available for inspection at each office of the firm and on the firm's website at www.winckworths.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Vonage
Deon, When you say I've tested up to 6 inbound calls at the same time with Broadvoice, is this with 6 $19.95 DID numbers that you have assigned to *? thanks Doug Deon Rodden wrote: When I initially signed up with Packet8 and they sent their converter, I used a X100P card in my Asterisk server so that it could send and receive calls through Packet8, I suspect the same trick would work for Vonage. The benefit is you can then have several phones in the house, or one at work, or a softphone in your laptop, register with your Asterisk server and then you can place and receive unlimited local/long distance calls through your Vonage account. You can also have Asterisk answer and you can use it's IVR/Automated Attendant functionalities. You will be limited to only 1 inbound/outbound call at a time though. I eventually canned Packet8 in favor of BroadVoice, $19.95 a month, unlimited local and long distance, and I've tested up to 6 inbound calls at the same time and it worked. Ask Bjørn Hansen wrote: On Aug 24, 2004, at 4:17 PM, Chris Shaw wrote: I hold no ill will towards Vonage but I have to say honestly... ewww... They've already made their feelings quite clear by refusing to allow people to bring their own devices and taking steps to even hide their SIP servers (changing the port from the RFC standard 5060 to 5061 for example.) Why not go with someone who's actually willing to allow you to use Asterisk and any phone you want like NuFone, BroadVoice, IconnectHere or a host of others instead of trying to hack Vonage... At least when I signed up with Vonage they were the only VoIP provider that had numbers in my old rate center and could transfer the number from SBC. It does, of course, suck not to be able to use it with Asterisk. (I could sign up for a soft-phone, but I don't think it'd be with my old number defeating the purpose...) - ask ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip change?
Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial(SIP/3008-9a9b, SIP/3000|15) in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2(SIP/3008-9a9b, u3000) in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandatory config change, or is sip broken? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Can't flash 7960: P0S30200 .bin not found
Lol. Known issue, I spent an hour working on that problem. The phone's current firmware is too hold and does not support longer filenames like that. You have to increment the firmware versions, 2 or 3 firmware upgrades and you'll be ready to use the latest and greatest. Try upgrading to P0S30203.bin first. Apparently Cisco has some kind of filename checking thing, as when I tried to rename P0S3-05-3-00.bin to P0S30503.bin it didn't take, neither did P0S3-06-3-00.bin to P0S30603.bin First I went to P0S30203 and then I went to P0S3-05-3-00 and then I went to P0S3-06-3-00 and then finally to P003-07-1-00 (Which is some kind of boot loader that then loads the SIP firmware, strange how 7-1 does it). However, I now have a phone that has an even older firmware, one that won't even take P0S30203.bin like the other one did. I'm reading and I think I need P0S30200.bin or P0S30200.bin ; I may even end up upgrading the Skinny/SCCP firmware a few versions before it'll jump to the SIP firmware. Pavel Jezek wrote: look to your SIPDefault.cnf or SIPmacaddress.cnf on TFTP server if you have is correct file name in image_version: section PJ - Original Message - From: Tracy R Reed Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Friday, August 27, 2004 11:48 AM Subject: Can't flash 7960: P0S30200 .bin not found When I try to flash my 7960 with SIP I get messages like this in the tftp server logfile: Aug 27 02:01:17 home tftpd[32590]: tftpd: trying to get file: P0S3-03-0-00 .bin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Vonage
No. Just one regular $19.95 residential plan. I've had 6 cell phones call my DID and my IVR picked up all 6 times. I never got a 7th cell, so I never tested the limit. But I don't want to abuse my BroadVoice account so I haven't tried it again. I mainly stick to 1 line, an occassional 2nd line/channel may be used, but I know it can do more. The way I interact with BroadVoice though isn't officially sanctioned, I didn't prefer to use their Asterisk Only SIP gateway, in which they charge you 3.2 cents a minute (or whatever) when you exceed the first line. Doug Shubert wrote: Deon, When you say I've tested up to 6 inbound calls at the same time with Broadvoice, is this with 6 $19.95 DID numbers that you have assigned to *? thanks Doug Deon Rodden wrote: When I initially signed up with Packet8 and they sent their converter, I used a X100P card in my Asterisk server so that it could send and receive calls through Packet8, I suspect the same trick would work for Vonage. The benefit is you can then have several phones in the house, or one at work, or a softphone in your laptop, register with your Asterisk server and then you can place and receive unlimited local/long distance calls through your Vonage account. You can also have Asterisk answer and you can use it's IVR/Automated Attendant functionalities. You will be limited to only 1 inbound/outbound call at a time though. I eventually canned Packet8 in favor of BroadVoice, $19.95 a month, unlimited local and long distance, and I've tested up to 6 inbound calls at the same time and it worked. Ask Bjørn Hansen wrote: On Aug 24, 2004, at 4:17 PM, Chris Shaw wrote: I hold no ill will towards Vonage but I have to say honestly... ewww... They've already made their feelings quite clear by refusing to allow people to bring their own devices and taking steps to even hide their SIP servers (changing the port from the RFC standard 5060 to 5061 for example.) Why not go with someone who's actually willing to allow you to use Asterisk and any phone you want like NuFone, BroadVoice, IconnectHere or a host of others instead of trying to hack Vonage... At least when I signed up with Vonage they were the only VoIP provider that had numbers in my old rate center and could transfer the number from SBC. It does, of course, suck not to be able to use it with Asterisk. (I could sign up for a soft-phone, but I don't think it'd be with my old number defeating the purpose...) - ask ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
OT re: [Asterisk-Users] sip change?
Kind of off topic but I know CVS is the prefered way of upgrading, however are there such things as stable CVS upgrades? It seems a lot of the CVS's have a lot of devel bugs in this that I would be scared to put even near production. Just IMHO. :-) Matt -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Friday, August 27, 2004 9:15 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] sip change? Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial(SIP/3008-9a9b, SIP/3000|15) in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2(SIP/3008-9a9b, u3000) in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandatory config change, or is sip broken? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip change?
Whenever I see the Maximum retries message it usually indicated a communication problem, like one way traffic. Last time I got it, I traced it to a bad firewall rule, dropped the firewall and it worked, the time before that when I received it, it was due to a routing error, the server could get the request but couldn't respond (wrong gateway). Rich Adamson wrote: Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial(SIP/3008-9a9b, SIP/3000|15) in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2(SIP/3008-9a9b, u3000) in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandatory config change, or is sip broken? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using regular expression in dialplan
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 = OK, Goto 4
else
Goto 2
I've tried the follwoing syntax but it is alway going to 2 whatever
the value of DTMFSeq:
exten = s,1,GotoIf($[${DTMFSeq} : 123]?4:2)
exten = s,2,SetVar(InvalidCount=$[${InvalidCount} + 1])
.
exten = s,4,SetVar(Result=ok)
The only way I managed to make it work is the following :
exten = s,1,GotoIf($[${DTMFSeq} : 1|2|3]?s|4:s|2)
But I'm not totaly satisfied with it as I'm going to check more
complex regex later ...
Thank you for your help
___
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Re: [Asterisk-Users] sip change?
* and the 7960's are on the same wire, no firewall involved whatsoever. Backing out to July 12th now... Whenever I see the Maximum retries message it usually indicated a communication problem, like one way traffic. Last time I got it, I traced it to a bad firewall rule, dropped the firewall and it worked, the time before that when I received it, it was due to a routing error, the server could get the request but couldn't respond (wrong gateway). Rich Adamson wrote: Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial(SIP/3008-9a9b, SIP/3000|15) in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2(SIP/3008-9a9b, u3000) in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandatory config change, or is sip broken? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Vonage
ok.. could we add a 'hunt group' to * and roll incoming calls over to several extensions? We also signed up with the Broadvoice 'BYOD' $19.95 service just in the past week and found the service to work extremely well with Asterisk. I also updated the * server to 1.0-RC2 before testing it. Doug Deon Rodden wrote: No. Just one regular $19.95 residential plan. I've had 6 cell phones call my DID and my IVR picked up all 6 times. I never got a 7th cell, so I never tested the limit. But I don't want to abuse my BroadVoice account so I haven't tried it again. I mainly stick to 1 line, an occassional 2nd line/channel may be used, but I know it can do more. The way I interact with BroadVoice though isn't officially sanctioned, I didn't prefer to use their Asterisk Only SIP gateway, in which they charge you 3.2 cents a minute (or whatever) when you exceed the first line. Doug Shubert wrote: Deon, When you say I've tested up to 6 inbound calls at the same time with Broadvoice, is this with 6 $19.95 DID numbers that you have assigned to *? thanks Doug Deon Rodden wrote: When I initially signed up with Packet8 and they sent their converter, I used a X100P card in my Asterisk server so that it could send and receive calls through Packet8, I suspect the same trick would work for Vonage. The benefit is you can then have several phones in the house, or one at work, or a softphone in your laptop, register with your Asterisk server and then you can place and receive unlimited local/long distance calls through your Vonage account. You can also have Asterisk answer and you can use it's IVR/Automated Attendant functionalities. You will be limited to only 1 inbound/outbound call at a time though. I eventually canned Packet8 in favor of BroadVoice, $19.95 a month, unlimited local and long distance, and I've tested up to 6 inbound calls at the same time and it worked. Ask Bjørn Hansen wrote: On Aug 24, 2004, at 4:17 PM, Chris Shaw wrote: I hold no ill will towards Vonage but I have to say honestly... ewww... They've already made their feelings quite clear by refusing to allow people to bring their own devices and taking steps to even hide their SIP servers (changing the port from the RFC standard 5060 to 5061 for example.) Why not go with someone who's actually willing to allow you to use Asterisk and any phone you want like NuFone, BroadVoice, IconnectHere or a host of others instead of trying to hack Vonage... At least when I signed up with Vonage they were the only VoIP provider that had numbers in my old rate center and could transfer the number from SBC. It does, of course, suck not to be able to use it with Asterisk. (I could sign up for a soft-phone, but I don't think it'd be with my old number defeating the purpose...) - ask ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using regular expression in dialplan
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 = OK, Goto 4
else
Goto 2
I've tried the follwoing syntax but it is alway going to 2 whatever
the value of DTMFSeq:
exten = s,1,GotoIf($[${DTMFSeq} : 123]?4:2)
exten = s,2,SetVar(InvalidCount=$[${InvalidCount} + 1])
.
exten = s,4,SetVar(Result=ok)
The only way I managed to make it work is the following :
exten = s,1,GotoIf($[${DTMFSeq} : 1|2|3]?s|4:s|2)
But I'm not totaly satisfied with it as I'm going to check more
complex regex later ...
Thank you for your help
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[Asterisk-Users] Asterisk compatible E1 cards
After days of searching i've finally figured out that E1 lines in india use multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards which would work on R2. Also if anyone on this list is from INDIA and uses asterisk and E1 lines please let me know what type of cards (which vendor) do you use and what type of signally your E1 line uses. -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicetronix Segmentation Fault
Hi, I am using a voicetronix OpenLine4. I downloaded a recent asterisk CVS from voicetronix webpage but have had no luck to reduce echo on outgoing calls and for it not to crach on incoming calls. I dont think both problems are related though. Here is an output of what happens when a new call comes in and my voicetronix tries to pick it up and crashes asterisk: vpb/1-1: Event [0=[00] Ring] vpb/1-1: handle_notowned: mode=3, event[0][[00] Ring ]=[0] vpb/1-1: New call for context [pstn] Aug 27 09:06:11 WARNING[19475]: pbx.c:1868 ast_pbx_run: Channel 'vpb/1-1' sent into invalid extension 's' in context 'default', but no invalid handler == vpb/1-1: Hangup requested vpb/1-1: Setting state down CID record - start vpb/1-1: Flushing event [11]=[00] Ring Off == vpb/1-1: Hangup complete Restarting monitor Trying to reawake monitor Monitor restarted CID record - skipped 602.460051ms trailing ring CID record - recorded 1711.737009ms between rings Segmentation fault Any advice on how to correct this or the other problem would be appreciated. ;) Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 with Fedora 2 GCC 3.3
Hi, did anybody managed to compile h323 channel under Fedora 2? There's only gcc 3.3 and 3.4. Does h323 from * or opencall work with FC2 and gcc 3.3? Anybody had similiar problems? tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compatible E1 cards
It's not the card's fault, it's the lack of a software driver fault. R2 has a country dependent implementation. Some countries even have two incompatible standards internally. Em Sex 27 Ago 2004 11:03, Vikram Rangnekar escreveu: After days of searching i've finally figured out that E1 lines in india use multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards which would work on R2. Also if anyone on this list is from INDIA and uses asterisk and E1 lines please let me know what type of cards (which vendor) do you use and what type of signally your E1 line uses. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to fetch a call?
Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's einen Ruf heranholen. It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to my collegue's desc to answer at his phone). I assume there is a key code I could type at my phone to fetch that call to my phone. Maybe there is some mechanism to grant that permission to me in some conf.file. Thanks for any hints or key words, where I can find explained that feature! Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems dialing out with T100P and Adtran
I have a T100P card connected to an Adtran and then a T1.
I have added the following configurations to Asterisk...but, when I dial
9 and then a local phone number, it bounces between the dial tone and
silence and the *error* light on the Adtran blinks.
zaptel.conf
span=1,0,0,esf,b8zs
fxsks=1-8
loadzone=us
defaultzone=us
zapata.conf
[channels]
context=from-sip
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=2
txgain=2
group=1
channel = 1-7
extensions.conf
...
[from-sip]
ignorepat = 9
exten = _9NXXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _91XXXNXXX,1,Dial(Zap/g1/${EXTEN:1})
; generic phone extension
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,VoiceMail(u1001)
exten = 1001,102,VoiceMail(b1001)
exten = 1001,103,Hangu
...
sip.conf
...
[1001]
type=friend
username=1001
fromuser=1001
callerid=User Name 1001
host=dynamic
nat=yes
canreinvite=yes
dtmfmode=info
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
context=from-sip
secret=1001
I am using Grandstream BT101 phones, plugged into my LAN. I can dial
extension/phone to extension/phone in the office just fine. But, when I
dial *9* to get out, nothing happens. I don't get the dial tone back
after I dial 9, and if I dial 9 and the number and send the call...the
server runs through what looks like a connection to a Zap channel...I
don't get any noticable erros...but the call never makes it out. Once,
I dialed the number again, while the Adtran was flashing erros and the
dial tone was going in and out and it rang my cell phone...but then
immediately hung up and closed out the call?
I'm new to Asterisk...any help or insight would be much appreciated.
Cheers,
--
Shawn Parker
Network Administrator
Cumulus Broadcasting, LLC.
Columbia-Jefferson City, Missouri
1.573.449.4141
[EMAIL PROTECTED]
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[Asterisk-Users] Re: how to fetch a call?
In article [EMAIL PROTECTED], Roger Schreiter [EMAIL PROTECTED] wrote: Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's einen Ruf heranholen. It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to my collegue's desc to answer at his phone). I don't know whether it is implemented or not in Asterisk, but the feature is known in English as call pickup. mfg, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 - SCCP or SIP?
On my experience, you should go to SIP whenever possible. 7940/60 on SIP will do most if not all fuctions. Try the little chart on support hardware on chan-sccp.sourceforge.net Lethol - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: Fri, 27 Aug 2004 14:16:11 +0100 Subject: [Asterisk-Users] Cisco 7940 - SCCP or SIP? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems this is where some sort of black magic needs to be used as I cannot find any way of getting them to work which leads me to the main question Is it better to use chan_sccp or SIP? I know these phones can work in either mode I was just wandering which is the better format and which has the most functions implemented? Its a simple home environment that I am planning but it would be good to be able to use the softkeys to transfer calls and to pickup messages. Thanks in advance, Sam Kevin Walsh [EMAIL PROTECTED] wrote on 27/08/2004 13:59:09: Michael Manousos [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Michael Manousos [EMAIL PROTECTED] wrote: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. I would propose that after x lost packets, Asterisk should treat all further lost packets as CN. The proceeding x packets should be interpreted as RTP packet loss and run through the concealment routine. Well, no matter what kind of concealment algorithm is used, just the first one or two packets will be concealed. The rest losses will result in no-playback. No CN interpretation, just absolute silence. That's true - unless there's some logic to say that after x lost packets, the line state should switch to CN generation instead of silence. The line state would switch back once a fresh RTP packet is received. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. In this case, there is nothing to conceal anyway, as the last received data was a CN packet. In this case, the CN state should be continued until an RTP packet is received and the line state can be changed. Exactly. So the receiver, in case of no-receiption, should go back and see what was the last packet correctly received and act as I described above. Maintaining an audio state flag (CN/RTP) would be the key here. The difficult part to handle would be late or out-of-sequence RTP packets. These should be ironed out by the jitter buffer. Late, lost and juggled packets are to be expected when dealing with UDP. Actually this is not so difficult, if there is a jitter buffer. Right. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Winckworth Sherwood Solicitors and Parliamentary Agents DX 148400 WESTMINSTER 5 : 35 Great Peter Street, London SW1P 3LR Telephone 020 7593 5000 Fax 020 7593 5099 Confidentiality This email message and any attachments are confidential; they may be subject to legal professional privilege and are intended for the named recipient only. If you are not the named recipient, please return the message and enclosures immediately and delete them from your system. Caution Before advice received only by email (whether by attachment or otherwise) may be relied on, the authenticity of the communication must be verified by means independent of email. Regulation The firm is regulated by the Law Society. Partners A list of partners is available for inspection at each office of the firm and on the firm's website at www.winckworths.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
[Asterisk-Users] Re: Asterisk compatible E1 cards
+++ Marcelo Pacheco [27/08/04 11:06 -0300]: It's not the card's fault, it's the lack of a software driver fault. R2 has a country dependent implementation. Some countries even have two incompatible standards internally. Em Sex 27 Ago 2004 11:03, Vikram Rangnekar escreveu: After days of searching i've finally figured out that E1 lines in india use multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards which would work on R2. Also if anyone on this list is from INDIA and uses asterisk and E1 lines please let me know what type of cards (which vendor) do you use and what type of signally your E1 line uses. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'd volunteer to code the R2 support for asterisk if someone could provide me some documentation about it. -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card
What about if I want to call a Free World Dialup number from asterisk and play a number? Jason Williams wrote: On Thu, 26 Aug 2004 09:25:55 -0600, Andrew Elchuk [EMAIL PROTECTED] wrote: Is a sound card needed in order to playback some of the asterisk sounds in /var/lib/asterisk/sounds when dialing out with an X100P? Thanks. No Sound card is requied ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew Elchuk Technical Associate Cronus Technologies 248 - 111 Research Drive Saskatoon, SK S7N 2X8 Tel: (306) 652-5798 ext. 112 Fax: (306) 652-5799 Toll Free: 1-877-655-5798 http://www.cronustech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to fetch a call?
http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup http://www.voip-info.org/tiki-index.php?page=Asterisk%20channels On Fri, 27 Aug 2004 16:11:46 +0200, Roger Schreiter [EMAIL PROTECTED] wrote: Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's einen Ruf heranholen. It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to my collegue's desc to answer at his phone). I assume there is a key code I could type at my phone to fetch that call to my phone. Maybe there is some mechanism to grant that permission to me in some conf.file. Thanks for any hints or key words, where I can find explained that feature! Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to Vonage
FWIW, I've had Broadvoice running for two or three months now. Very reliable, good folks in tech-support helped with the initial asterisk config and getting me SIP credentials. Since my set-up as a home-pbx, incoming calls ring all my extensions (Sipuras) all the time. If someone's already talking, they hear a call-waiting beep (again, Sipura config) while the other extensions are rining. It's truly confusing to my wife, yet delightfully functional. For Vonage, I've kept my ATA186 to retain two number associated with it. I've added a $10/month softline and configured the hardline to simulring, so all Vonage calls come in via the softline now. This also has worked flawlessly for around two months now. I'm considering wasting a previous FXO port on the ATA186 to utilize the 500 outgoing minutes I have, but between the unlimited state-wide calling (from Broadvoice) and the 500 long-distance minutes on the softline (and simpletelecom's beta), we don't even need those minutes. Economically, I'm considering getting rid of Vonage entirely. They currently cost us $25/month, and that's just to get a certain area code that's unavailable elsewhere. For $25, we could receive a lot of incoming 800# calls... Someday. -Original Message- From: Doug Shubert [mailto:[EMAIL PROTECTED] Sent: Friday, August 27, 2004 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Vonage ok.. could we add a 'hunt group' to * and roll incoming calls over to several extensions? We also signed up with the Broadvoice 'BYOD' $19.95 service just in the past week and found the service to work extremely well with Asterisk. I also updated the * server to 1.0-RC2 before testing it. Doug Deon Rodden wrote: No. Just one regular $19.95 residential plan. I've had 6 cell phones call my DID and my IVR picked up all 6 times. I never got a 7th cell, so I never tested the limit. But I don't want to abuse my BroadVoice account so I haven't tried it again. I mainly stick to 1 line, an occassional 2nd line/channel may be used, but I know it can do more. The way I interact with BroadVoice though isn't officially sanctioned, I didn't prefer to use their Asterisk Only SIP gateway, in which they charge you 3.2 cents a minute (or whatever) when you exceed the first line. Doug Shubert wrote: Deon, When you say I've tested up to 6 inbound calls at the same time with Broadvoice, is this with 6 $19.95 DID numbers that you have assigned to *? thanks Doug Deon Rodden wrote: When I initially signed up with Packet8 and they sent their converter, I used a X100P card in my Asterisk server so that it could send and receive calls through Packet8, I suspect the same trick would work for Vonage. The benefit is you can then have several phones in the house, or one at work, or a softphone in your laptop, register with your Asterisk server and then you can place and receive unlimited local/long distance calls through your Vonage account. You can also have Asterisk answer and you can use it's IVR/Automated Attendant functionalities. You will be limited to only 1 inbound/outbound call at a time though. I eventually canned Packet8 in favor of BroadVoice, $19.95 a month, unlimited local and long distance, and I've tested up to 6 inbound calls at the same time and it worked. Ask Bjørn Hansen wrote: On Aug 24, 2004, at 4:17 PM, Chris Shaw wrote: I hold no ill will towards Vonage but I have to say honestly... ewww... They've already made their feelings quite clear by refusing to allow people to bring their own devices and taking steps to even hide their SIP servers (changing the port from the RFC standard 5060 to 5061 for example.) Why not go with someone who's actually willing to allow you to use Asterisk and any phone you want like NuFone, BroadVoice, IconnectHere or a host of others instead of trying to hack Vonage... At least when I signed up with Vonage they were the only VoIP provider that had numbers in my old rate center and could transfer the number from SBC. It does, of course, suck not to be able to use it with Asterisk. (I could sign up for a soft-phone, but I don't think it'd be with my old number defeating the purpose...) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk compatible E1 cards
First there's Analog R2 and digital R2. I'm concerned with digital R2 (R2D) only. R2 is equivalent to the american robbed bit signalling used in the US. Q.421 is the ITU document number. It costs money to download the specification. And it is only the generic part of it, it doesn't cover the country-to-country specifics. I have a document in Portuguese for the Brazilian R2D specification, meaning the specifics for our country. I'm intending to write it, but I'll only have the hardware about 1 month from now. For India, you would need to contact the Indian telephony regulatory agent, maybe they have it available over the Internet, I got the Brazilian document through our national agency's site. Also, before you can plug a Digium card to any Indian telco provided E1, you need to get the card certified, otherwise it can get you jail time/fines, at least in other countries. Here in Brazil I'm planning to plug it into a Brazilian certified PBX to avoid any legal trouble. Marcelo Pacheco Em Sex 27 Ago 2004 11:22, Vikram Rangnekar escreveu: +++ Marcelo Pacheco [27/08/04 11:06 -0300]: It's not the card's fault, it's the lack of a software driver fault. R2 has a country dependent implementation. Some countries even have two incompatible standards internally. Em Sex 27 Ago 2004 11:03, Vikram Rangnekar escreveu: After days of searching i've finally figured out that E1 lines in india use multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards which would work on R2. Also if anyone on this list is from INDIA and uses asterisk and E1 lines please let me know what type of cards (which vendor) do you use and what type of signally your E1 line uses. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sip change? (Rich Adamson)
Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter Message: 5 Date: Fri, 27 Aug 2004 08:45:19 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] sip change? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 * and the 7960's are on the same wire, no firewall involved whatsoever. Backing out to July 12th now... Whenever I see the Maximum retries message it usually indicated a communication problem, like one way traffic. Last time I got it, I traced it to a bad firewall rule, dropped the firewall and it worked, the time before that when I received it, it was due to a routing error, the server could get the request but couldn't respond (wrong gateway). Rich Adamson wrote: Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial(SIP/3008-9a9b, SIP/3000|15) in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2(SIP/3008-9a9b, u3000) in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandatory config change, or is sip broken? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ACD ringall + roundrobin
Hi all, I have a need where ACD ringall and ACD roundrobin ring strategies will be combined. Basically, ring every agent in a specified order, but whenever it times out and goes to the next agent, I still need the previous agent(s) to continue to ring. I would like to develop this extension myself as a contribution to the asterisk community. In doing this, where should I start? I've starting probing the code for ACD, but I'm wondering if I would be better off just waiting for ICD to become a little more mature, as the documentation for ICD states it is possible, via the config (?), to combine stategies? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Announcement not until after # accept call pressed
When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after which the call is immediately connected. Is there a way to have the announcement played to the agent before they press # to accept. I have ackcall=yes in agent.conf Can't find anything on the wiki. Thanks Andrew [exten.conf] exten = s,1,Answer exten = s,2,background(custom/100) ; Sales exten = 1,1,ringing(2) exten = 1,2,playback(custom/101) exten = 1,3,queue(sales) [queue.conf] [default] ; ; Default settings for queues (currently unused) ; [sales] music = default announce = sales_queue; This not played until after # pressed .. How can i get announce to play as soon as call answered? announce-frequency = 20 strategy = roundrobin timeout = 15 retry = 5 maxlen = 0 member = Agent/7001 member = Agent/7005 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Overhead Paging
Traditional overhead paging systems are a little more complicated than they first appear. It's not just speakers and a centralized amplifier. They would have too much cable loss if done that way. Instead, they use a centralized power source, and amplifiers at each speaker unit.. The one I just took apart used 3 wires in series to the overhead speakers, and each speaker unit had an independent volume control and an onboard transformer on the unit. One of the wires (black) carried 60V. Green was ground. The last wire (smaller than the rest) was the audio signal. The audio signal ran into the transformer and modulated the 60V signal, and the output went to the speaker. Neat contraptions. Don't hurt yourself. -ejay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw Sent: Thursday, August 26, 2004 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Overhead Paging - Original Message - From: Brian Pavane [EMAIL PROTECTED] To: Chris Shaw [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 2:39 PM Subject: Re: [Asterisk-Users] Overhead Paging Chris, What you're talking about is exactly what I'm looking for. I'm interested in the middleware that would sit between the speakers and the IAD. I have found the Bogen TAMB device -- however I was wondering if anyone had any experience with this box. How many speakers can you power off of the unit without needing an external amp, etc... -Brian lol those would be electrical questions... Not sure how to answer those, it would depend on power output of the TAMB device and how much current draw and wattage the speakers require... I know this would work though, as long as the receiver/amplifier has an FXO interface and not an RCA jack like you would find on a stereo system or CD player... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] xlite Problems
-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and asterisk crashes. I'm running the CVS from yesterday. Any ideas? Here's the sip.conf 1009 is identical: [101] type=friend callerid=Tim Jackson 100 host=dynamic dtmfmode=rfc2833 nat=no; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT context=default disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] auto-gain, or different gain between incoming and outgoing calls (EURO ISDN PRI) ?
Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain by about 4 - 7, but then all my other phones are so loud that it distorts and the echo-canceller can't compensate (on outgoing calls only). The ringing noise also is at full level (deafening loud). I have also noticed that incoming calls from PSTN into the TE405P to SIP are amplified differently than outgoing calls from SIP to PSTN. It seems the TXgain is not used on incoming calls... Are my observations above expected, or is there something wrong with the code? Is auto-gain implementable / recommendable so that all the SIP phones will sound the same (volume-wise) to the outside PSTN user, and vice versa ? Could we build this into the sip configuration so that individual gain per phone is adjustable if needed ? Here is a cut-out of my zapata.conf (just in case I am really stupid :-) [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=yes cancallforward=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=5.0 txgain=7.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 context=default signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 context=ppms signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 context=default signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 context=default signalling=pri_net channel = 94-108 channel = 110-124 Hope anyone can shed some light on this. I have been breaking my head on this about 4 days now, trying just about anything... Thanks Walter Klomp Singapore. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: sip change? (Rich Adamson)
Interesting... but that was supposed to have been a fix for Uniden bugs. It shouldn't have negatively impacted 7960's on the same wire. Must be a broken logic in there somewhere. Rich Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter Message: 5 Date: Fri, 27 Aug 2004 08:45:19 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] sip change? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 * and the 7960's are on the same wire, no firewall involved whatsoever. Backing out to July 12th now... Whenever I see the Maximum retries message it usually indicated a communication problem, like one way traffic. Last time I got it, I traced it to a bad firewall rule, dropped the firewall and it worked, the time before that when I received it, it was due to a routing error, the server could get the request but couldn't respond (wrong gateway). Rich Adamson wrote: Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial(SIP/3008-9a9b, SIP/3000|15) in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2(SIP/3008-9a9b, u3000) in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandatory config change, or is sip broken? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice User hung up on voicemail
After a call is sent to voicemail on an inbound connection from Broadvoice, the call is hung up in the middle of recording a voice mail after about 30 or so seconds. I get an error User hung up. If I answer the call and not have it go to voicemail, the call will stay connected. This only seems to happen on the Broadvoice connection and voicemail. Is anyone experiencing this issue or able to resolve? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem dialing out to Free World Dialup
Hi I am trying to make a call to a Free World Dialup number with the following call file: Channel: SIP/[EMAIL PROTECTED] Callerid: Nagios MaxRetries: 0 WaitTime: 30 Context: autodialout Extension: s Priority: 1 When I put the file in /var/spool/asterisk/outgoing/ directory, the X-Lite software phone installed on my other computer rings once then it says Hung up in the window of the phone. This is output I get from the CLI: -- Attempting call on SIP/[EMAIL PROTECTED]:5060 for [EMAIL PROTECTED]:1 (Retry 1) Aug 27 09:20:22 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Aug 27 09:20:22 NOTICE[1200884528]: pbx_spool.c:235 attempt_thread: Call failed to go through, reason 1 Aug 27 09:20:28 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Any help as to why it rings once then stops would be greatly appreciated, thanks. -- Andrew Elchuk Technical Associate Cronus Technologies 248 - 111 Research Drive Saskatoon, SK S7N 2X8 Tel: (306) 652-5798 ext. 112 Fax: (306) 652-5799 Toll Free: 1-877-655-5798 http://www.cronustech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Vonage
The way I interact with BroadVoice though isn't officially sanctioned, I didn't prefer to use their Asterisk Only SIP gateway, in which they charge you 3.2 cents a minute (or whatever) when you exceed the first line. Where did you get this info? I have been using broadvoice for 2 months now and have never heard of this? I have heard of the 3.2cents a minute thing, but have never experienced it myself, I occasionally have calls on more than one line when others are using the phone and I don't know it... ok.. could we add a 'hunt group' to * and roll incoming calls over to several extensions? This totally defeats the purpose of VoIP This is going back to circuit-switched mentality... Remember that in VoIP a Line is just a username assigned to you by an ITSP, it can be a name or a number... You don't need rollover because it's just a connection like someone sending an E-Mail to your SMTP server... I realize what you mean, getting several accounts and Rolling them over so you can have multiple call appearances, but this breaks the whole idea of a pure VoIP setup... At $20-30 a month, you might as well use a TDM400P and add a second PSTN line, there are plans out there where when you sign up for a 2nd line you get unlimited long distance... My 0.0002 -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Release 1.01 of FWD Assistant available (bugfix release)
Hi (B (Ba bugfix release is now available for the FWD Assistant (B... (B (Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+FWD+Assistant (B (Bfurther details are on the Wiki. (B (Bthanks to everybody who has provided feedback (B (Brgds (Bbenjk (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem dialing out to Free World Dialup
On Fri, 27 Aug 2004, Andrew Elchuk wrote: -- Attempting call on SIP/[EMAIL PROTECTED]:5060 for [EMAIL PROTECTED]:1 (Retry 1) Aug 27 09:20:22 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Aug 27 09:20:22 NOTICE[1200884528]: pbx_spool.c:235 attempt_thread: Call failed to go through, reason 1 Aug 27 09:20:28 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Any help as to why it rings once then stops would be greatly appreciated, thanks. Usually this sort of stuff is a symptom of connectivity/NAT problems between the two SIP endpoints. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] questions and recommendations
Hi Yawl, After about 6 months of prattting about I've convinced my boss that we should be installing * into our currently under constuction Data Center in Somerset NJ. There will be 10 permanent people and DR space for another 50. My plan is as follows; ATAComm dual XEON server with quad T1 board. A handfull of ATA's for fax machines, job lot of X-Pro softphones for the DR bit, Polycom IP conference phone and a bunch of SNOM 200 phones for the permanent staff. Connection to the main office will be via SIP to a Cisco router we already have connected to the main office Lucent PBX. Does this sound like a runner? Comments please. Also, I'm having a problem trying to work out how to configure a group pickup line. I know I can add the pickupgroup line in sip.conf but that only allows me to do *8 when someone elses phone rings. What I want to do is have a button on every member phone that lights up when line 5 rings. Then if the call is for Fred rather than me, Fred can press the line 5 button and take the call. This is working on our Lucent currently Thanks folks. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice User hung up on voicemail
Yep... It's BroadVoice's problem, not *'s... When * is recording, be it voicemail or the record() application, * does not transmit a single packet back to BroadVoice (Confirmed by ethereal and TCPDump) After 30 seconds the BroadVoice switch will disconnect the call believing that it's a far-end disconnect... I think that once CNG is implemented in *, this problem should be fixed, but until then, you get 30 seconds of recording... period :( -Chris - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 27, 2004 8:20 AM Subject: [Asterisk-Users] Broadvoice User hung up on voicemail After a call is sent to voicemail on an inbound connection from Broadvoice, the call is hung up in the middle of recording a voice mail after about 30 or so seconds. I get an error User hung up. If I answer the call and not have it go to voicemail, the call will stay connected. This only seems to happen on the Broadvoice connection and voicemail. Is anyone experiencing this issue or able to resolve? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 SIP Firmware - Help.
Hi all, hope this isn't a duplicate - but my first post went AWOL. Sorry for this cheeky request and one that will probably not meet with much response but I may as well ask! I've just recieved my Cisco 7940 and am after upgrading it to the SIP firmware. I don't (yet) have a support contract and thus can't download the firmware image from the Cisco site. I will be getting a support contract as soon as my reseller sorts it for me but had quite wanted to get this phone working at the weekend (which in the UK is a three day weekend with a bank holiday monday). Would anyone be prepared to mail me off list with a copy of the firmware image for my weekend play, and I *promise* to obtain a support contract ASAP! Cheeky but sincere! ;-) Many thanks, Benjamin -- Benjamin Johnson Director thinktech Ltd. - Appropriate security solutions for business. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 -- IAX2 confusion, it doesn't work...
I am trying to get two * boxes to communicate with eachother.
I have read
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
as well as information on IAX channels, the Dial() command, and the switch
statement in extensions.conf.
But I am having no luck. I have a working * box running with a Zap card that
I want to be the server machine. I have another little box running * with
just a single SIP phone attached that I want to be the slave machine.
I am trying to get to where I can dial an extension on the SIP phone and have
it connect to the master * box and dial an extension there in the internal
context.
As per the dual+servers document, I have the following in the iax.conf on the
side with the SIP phone (the side to dial out of):
register = asterisk:[EMAIL PROTECTED]
...
[MainServer]
type=user
secret=lilbuddy
context=internal
I have no port= set because I want them both to default to the IAX2 port.
On the side with the TDM card, where I want to call from the SIP phone to, I
have the following in the iax.conf file:
[asterisk]
type=peer
context=internal
secret=lilbuddy
host=dynamic
dual+servers then goes into an example that I cannot comprehend:
[default]
exten = _801XXX,1,Goto,left|${EXTEN}|1
exten = _802XXX,1,Goto,right|${EXTEN}|1
[left]
exten = _801XXX,1,StripMSD,3
exten = _XXX,2,Goto,1
switch = IAX/left
[right]
exten = _802XXX,1,StripMSD,3
exten = _XXX,2,Goto,1
switch = IAX/left
I can see that if a call matches 801... or 802... it will go to the left or
right contexts respectively. And the first thing it does there is strip off
the first three digits and goes to the resulting extension. That takes us to
the Goto(1). Where does that go? Does the switch = statement do the same
thing as an include, but it hops to another server? And in this case, what
does IAX/left mean? and why is it included in *both* left and right?
The explanation in the wiki page for extensions.conf is as confusing:
[iaxprovider]
switch = IAX2/user:[EMAIL PROTECTED]/context
What exactly does this do? There are no extensions and it's not clear to me
if this is to be included into another context or seomthing.
So, looking at other pages in the wiki, I have decided to try to just use the
Dial() commant to reach over to the main * box (the one I want to call to).
So on the box with the SIP phone, I have the default context for the SIP phone
with this as the only entry:
exten = _XX,1,Dial(IAX2/asterisk:[EMAIL PROTECTED]/[EMAIL PROTECTED])
Which should call 192.168.0.250, go into the internal context, to the given
extension. On the Master, in the internal context, I have extension 22 to
ring my desk phone. This has been tested and works.
So what should happen is that I pick up the SIP phone, dial 22 and it will
execute the Dial(), login to 192.168.0.250, extension 22 in the internal
context, and ring my desk phone.
What happens instead (starting from a CLI invokation of asterisk -vvvc on
each machine) is:
The master loads all it's configuration and gives me: *CLI
I start the little slave box, and I get:
The slave loads all its configuration and I get *CLI. The master does
not indicate a registration at all of the slave, but iax.conf on the slave
indicates to register.
So here we sit. I pick up the SIP phone and dial 22 and on the slave (to
which the SIP phone is connected) I get:
*CLI -- Executing Dial(SIP/grandstream1-c62b, IAX2/asterisk:[EMAIL
PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called asterisk:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Aug 27 11:40:30 WARNING[131080]: chan_iax2.c:5352 socket_read: Call rejected by
192.168.0.250: No authority found
-- Hungup 'IAX2/192.168.0.250:4569/2'
== No one is available to answer at this time
So I can see that it executed the dial as it should have, and that the id and
secret match that in the iax.conf file on the main server.
The output on the master is:
*CLI Aug 27 11:30:36 NOTICE[131080]: chan_iax2.c:5251 socket_read: Rejected connect
attempt from 192.168.0.147
So it did reject the connection, but I'm not
[Asterisk-Users] Cisco 7940 Sip Firmware
Hi all, a cheeky request and one that will probably not meet with much response but I may as well ask! I've just recieved my Cisco 7940 and am after upgrading it to the SIP firmware. I don't (yet) have a support contract and thus can't download the firmware image from the Cisco site. I will be getting a support contract as soon as my reseller sorts it for me but had quite wanted to get this phone working at the weekend (which in the UK is a three day weekend with a bank holiday monday). Would anyone be prepared to mail me off list with a copy of the firmware image for my weekend play, and I *promise* to obtain a support contract ASAP! Cheeky but sincere! ;-) Many thanks, Benjamin -- Benjamin Johnson Director thinktech Ltd. - Appropriate security solutions for business. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXOs
Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least to me) although several in the user community make refernce to new firmware that might address the issue, real soon now. I see a lot of activity recently on-list about the TDM-400. Of course, mentions on-list are more than likely the result of people having problems. We don't hear about people who have no issues with a product. So, the nature of my inquiry is to explore how many people out here have good/great experiences with the various small FXO adapters? While the TDM-400 is my next possible purchase I'd also like to hear about devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so many products being offered I would hope that we have some collective experience with each one. Thanks, Michael Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713)861-4005 o(800)905-6412 f(713)864-8668 c(713)201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Are there any graphic designers on this list?
Hi (B (BI had asked for some help with the Asterisk Assistants (B (Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX (B (Band many have offered assistance with translations which I (Bam grateful for and like to say thank you again. (B (BHowever, there hasn't been a single response from a (Bgraphic designer to offer help with a custom icon. Are (Bthere any graphic designers on this list at all? If so, (Bplease take a look at the Wiki above and see if you can (Bhelp. (B (Bthanks (Brgds (Bbenjk (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie needs help - Dev_Kit_Lite installationproblem
Thanks Don for the help but I found someone with a similar problem and what they did was remove the audio module with a simple rmmod audio. Now i get dial tone and everything from my both both my x100p and my s100u. And can dial out from both. Thanx again Dave - Original Message - From: David Luong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 2:35 PM Subject: [Asterisk-Users] Newbie needs help - Dev_Kit_Lite installationproblem Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I too had the DevKitLite hardware. Had nothing but problems with it. Is difficult to get running and when you do get it going, I think you will find that you may have trouble getting it to dial correctly. After getting it running, I would pick up the phone to make a call and I would get a dial tone but the S100U would not accept the DTMF tones for dialing. The only thing I could do to correct this would be to down * and reboot the computer. Finally I stopped using it and bought a one port TDM400U. It works with no problems. I used the sample configuration files from digium documentaion that was supposed to be sane defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here is my probelm: This is what i did. unplugged s100u rmmod wcfxo rmmod wcusb rmmod zaptel replugged s100u modprobe wcfxo modprobe wcusb ztcfg -vv asterisk -cv This is what I got: [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on channel 2: No such device or address (6) /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# modprobe wcusb [EMAIL PROTECTED] root]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) [EMAIL PROTECTED] root]# asterisk -cv Asterisk CVS-HEAD-08/24/04-09:05:32, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] [Answer] [BackGround] [Busy] [Congestion] [DigitTimeout] [Goto] [GotoIf] [GotoIfTime] [Hangup] [NoOp] [Prefix] [Progress] [ResetCDR] [ResponseTimeout] [Ringing] [SayNumber] [SayDigits] [SayAlpha] [SayPhonetic] [SetAccount] [SetAMAFlags] [SetGlobalVar] [SetLanguage] [SetVar] [StripMSD] [Suffix] [Wait] [WaitExten] Asterisk Dynamic Loader Starting: [chan_modem.so] = (Generic Voice Modem Driver) = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) [res_musiconhold.so] = (Music On Hold Resource) [res_adsi.so] = (ADSI Resource) [res_features.so] = (Call Parking Resource) [res_crypto.so] = (Cryptographic Digital Signatures) [res_indications.so] = (Indications Configuration) [res_monitor.so] = (Call Monitoring Resource) [res_agi.so] = (Asterisk Gateway Interface (AGI)) [chan_sip.so] -z: No such file or directory = (Session Initiation Protocol (SIP)) [chan_modem_bestdata.so] = (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver) [chan_agent.so] = (Agent Proxy Channel) [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) [chan_local.so] = (Local Proxy Channel) [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) Aug 26 15:12:15 WARNING[1076220544]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled [chan_oss.so] = (OSS Console Channel Driver) Aug 26 15:12:15 WARNING[1076220544]: chan_oss.c:992 load_module: XXX I don't work right with non-full duplex sound cards XXX Aug 26 15:12:15 WARNING[1097410752]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_phone.so] = (Linux Telephony API Support) [chan_zap.so] = (Zapata Telephony w/PRI) Aug 26 15:12:16 WARNING[1076220544]: chan_zap.c:721 zt_open: Unable to specify channel 2: Device or resource busy Aug 26 15:12:16 ERROR[1076220544]: chan_zap.c:5869 mkintf: Unable to open channel 2: Device or resource busy here = 0, tmp-channel = 2, channel = 2 Aug 26 15:12:16 ERROR[1076220544]: chan_zap.c:8776 setup_zap: Unable to register channel '2' Aug 26 15:12:16 WARNING[1076220544]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 -- Unregistered channel 1 Aug 26 15:12:16 WARNING[1076220544]: loader.c:423 load_modules: Loading module chan_zap.so
[Asterisk-Users] Can a Macro call another Macro ?
Stupid newbie question that has probably been answered before but can a Macro call another Macro ??? is there any rules about how deep ??? Gary G. Hendershot Chief Technical Officer Advanced Digital Technologies BEGIN:VCARD VERSION:2.1 N:Hendershot;Gary FN:Gary Hendershot ([EMAIL PROTECTED]) ORG:Advanced Digital Technologies TITLE:Chief Technical Officer TEL;WORK;VOICE:(703) 280-2703 TEL;WORK;FAX:(703) 280-0979 ADR;WORK:;;2705 Elsemore Street;Fairfax;VA;22031;United States of America LABEL;WORK;ENCODING=QUOTED-PRINTABLE:2705 Elsemore Street=0D=0AFairfax, VA 22031=0D=0AUnited States of America URL;WORK:http://www.advdigtech.com EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20040604T134123Z END:VCARD ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there any graphic designers on this list?
ooohhh I'll take a crack at it! sounds like fun! :) (B- Original Message - (BFrom: "Sunrise Ltd" [EMAIL PROTECTED] (BTo: "astusr" [EMAIL PROTECTED] (BSent: Friday, August 27, 2004 8:47 AM (BSubject: [Asterisk-Users] Are there any graphic designers on this list? (B (B (B Hi (B (B I had asked for some help with the Asterisk Assistants (B (B (Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX (B (B and many have offered assistance with translations which I (B am grateful for and like to say thank you again. (B (B However, there hasn't been a single response from a (B graphic designer to offer help with a custom icon. Are (B there any graphic designers on this list at all? If so, (B please take a look at the Wiki above and see if you can (B help. (B (B thanks (B rgds (B benjk (B (B -- (B Sunrise Telephone Systems Ltd (B 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B __ (B GANBARE! NIPPON! (B Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (B http://mail.ganbare-nippon.yahoo.co.jp/ (B (B ___ (B Asterisk-Users mailing list (B [EMAIL PROTECTED] (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging
... Instead, they use a centralized power source, and amplifiers at each speaker unit... I've never seen one like that, That would suck nut bigtime... Especially if you had a large building with 100+ speakers, you would have to tune the volume on each one... That's not how the TAMB works, it has a centralised volume control I think he was describing a PA system that has either lots of speakers or very long cable runs. It use to be rather popular in those cases to use an amplifier that produced higher voltage audio (and therefore the need for lower current per speaker) to avoid the loss of power to each speaker due to the cable runs, etc. In that case, a step-down transformer was required at each speaker (or room), and if the customer wanted room volume controls, then an additional control had to be wired to the speaker. I would seriously doubt whether folks see much of that anymore. Lots of other ways to address boat-loads of speakers and long cable runs with current technology. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GRSecurity and ALSA on a Gentoo Server
Deon Rodden wrote: I've been working with Asterisk for about 2 months now and am doing well. However I decided to switch platforms from Fedora Core 1, that my predacessor was using, to Gentoo, for obvious reasons. It just seems faster and less bloated everything I need, nothing I don't. Anyways, I've read what the Wiki had to say about it and I was only confused on one thing, putting ALSA in my USE statement. It's a 1U server with no Sound Card. I did not choose to put ALSA in my USE flags as I don't have a sound card. But will Asterisk suffer in any way? I know that Asterisk is fully capable of running on a machine with No Sound card, my Fedora servers have no sound card, but by ommitting alsa in my USE flags, will Asterisk be compiled in a way that would make it less functional? No. There is no problem installing or running it with USE=-alsa. My last question, sorry guys (and girls), is about the grsecurity in the 2.4 kernel (I chose 2.4 instead of 2.6). I set it to low for now, as it said it wouldn't cause any compatibility issues with 99% of the programs. Has anybody tried medium, or even high, with Asterisk? How secure can you get the kernel without interfering with Asterisk. Yes, I use asterisk with grsec on high. No problems. This is just more of a comment, but if anybody see's anything wrong with it I'd like to know. I don't want to use the 0.9.0 ebuild (but I emerged it just to get the dependencies taken care of) so I emerge'd the CVS program so that I can upgrade libpri/zaptel/asterisk from 0.9.0 to the latest. The The Wiki mentions something about CVS and points to: http://bugs.gentoo.org/show_bug.cgi?id=33345 but that link is dead. I figured I'd just CVS Asterisk the normal way, do the make install and it should upgrade it. I don't use the portage ebuild for Asterisk, so I don't really know. However, after a brief look at the ebuild, it looks like everything is in the right place. To be sure, though, I would 'emerge unmerge asterisk zaptel libpri' and install these fresh using the normal configure/make/make install methods from the CVS sources. Regards, Deon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems dialing out with T100P and Adtran
Nevermind. I was a digit off in my zaptel.conf...
the span for my adtran settings is 1,1,0,esf,b8zs instead of the one i
have listed below.
ph...one digit off.
cheers,
Shawn Parker wrote:
I have a T100P card connected to an Adtran and then a T1.
I have added the following configurations to Asterisk...but, when I
dial 9 and then a local phone number, it bounces between the dial tone
and silence and the *error* light on the Adtran blinks.
zaptel.conf
span=1,0,0,esf,b8zs
fxsks=1-8
loadzone=us
defaultzone=us
zapata.conf
[channels]
context=from-sip
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=2
txgain=2
group=1
channel = 1-7
extensions.conf
...
[from-sip]
ignorepat = 9
exten = _9NXXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _91XXXNXXX,1,Dial(Zap/g1/${EXTEN:1})
; generic phone extension
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,VoiceMail(u1001)
exten = 1001,102,VoiceMail(b1001)
exten = 1001,103,Hangu
...
sip.conf
...
[1001]
type=friend
username=1001
fromuser=1001
callerid=User Name 1001
host=dynamic
nat=yes
canreinvite=yes
dtmfmode=info
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
context=from-sip
secret=1001
I am using Grandstream BT101 phones, plugged into my LAN. I can dial
extension/phone to extension/phone in the office just fine. But, when
I dial *9* to get out, nothing happens. I don't get the dial tone
back after I dial 9, and if I dial 9 and the number and send the
call...the server runs through what looks like a connection to a Zap
channel...I don't get any noticable erros...but the call never makes
it out. Once, I dialed the number again, while the Adtran was
flashing erros and the dial tone was going in and out and it rang my
cell phone...but then immediately hung up and closed out the call?
I'm new to Asterisk...any help or insight would be much appreciated.
Cheers,
--
Shawn Parker
Network Administrator
Cumulus Broadcasting, LLC.
Columbia-Jefferson City, Missouri
1.573.449.4141
[EMAIL PROTECTED]
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Re: [Asterisk-Users] FXOs
Hi, I have only done some basic testing with the sipaura 3000 that I bought 2 weeks ago - it appears to work very well. I have made quite a number of calls - although they have only been short ones and not had a single echo yet. No experience yet with any other devices but would like to hear about peoples experiences with ISDN BRI adapters Cheers, Sam [EMAIL PROTECTED] wrote on 27/08/2004 16:41:12: Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least to me) although several in the user community make refernce to new firmware that might address the issue, real soon now. I see a lot of activity recently on-list about the TDM-400. Of course, mentions on-list are more than likely the result of people having problems. We don't hear about people who have no issues with a product. So, the nature of my inquiry is to explore how many people out here have good/great experiences with the various small FXO adapters? While the TDM-400 is my next possible purchase I'd also like to hear about devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so many products being offered I would hope that we have some collective experience with each one. Thanks, Michael Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713)861-4005 o(800)905-6412 f(713)864-8668 c(713)201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Winckworth Sherwood Solicitors and Parliamentary Agents DX 148400 WESTMINSTER 5 : 35 Great Peter Street, London SW1P 3LR Telephone 020 7593 5000 Fax 020 7593 5099 Confidentiality This email message and any attachments are confidential; they may be subject to legal professional privilege and are intended for the named recipient only. If you are not the named recipient, please return the message and enclosures immediately and delete them from your system. Caution Before advice received only by email (whether by attachment or otherwise) may be relied on, the authenticity of the communication must be verified by means independent of email. Regulation The firm is regulated by the Law Society. Partners A list of partners is available for inspection at each office of the firm and on the firm's website at www.winckworths.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No audio on PRI channel answered by Playback() or MeetMe()
Does Asterisk need a sound card or
functional Console/dsp to answer inbound DID number from PRI and playback
.gsm files?I can call from any of the SIP extensions on Asterisk and
hear audio from Playback(), MeetMe(), or MOH. The problem I am having
with calls from my PRI is as follows:I have an Asterisk
(CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with
PRI. I have a single DID number that rings in from the NEC IPX on PRI
Span 1, trunk group 1. If I assign the inbound DID to ring an
extension on Asterisk, ie. SIP/2000, it in fact rings and when answered I
have a complete 2-way voice path. If I change the destination of the
inbound DID from SIP/2000 to MeetMe() or Playback(), Asterisk will answer
and I can see from the CLI the .gsm file being played but there is no
playback audio heard on the calling extension.If I assign the DID to
ring extension SIP/2000 and then after time-out send it to MeetMe() or
Playback() it works and the caller hears the .gsm file. Any assistance in
solving this problem is appreciated.What follows are two examples from
what I tried in extensions.conf:This works but is not
desirable:[nec_pri]; Digital PRI from the NEAX2400exten
= 2688,1,Wait,1exten = 2688,2,Dial(SIP/2000,3,Tr)exten =
2688,3,Wait,1exten = 2688,4,MeetMe,|Mpsexten =
2688,5,HangupThis will answer, but there is no audible playback on the
channel:[nec_pri]; Digital PRI from the NEAX2400exten =
2688,1,Wait,3exten = 2688,2,MeetMe,|Mpsexten =
2688,3,HangupThis is what is displayed from the CLI while the calling
station is connected via PRI: -- Accepting call from
'2502' to '2688' on channel 0/4, span 1 -- Executing
Wait("Zap/4-1", "3") in new stack -- Executing MeetMe("Zap/4-1",
"|Mps") in new stack -- Playing 'conf-getconfno' (language
'en') -- Playing 'conf-getconfno' (language
'en') -- Playing 'conf-getconfno' (language
'en') -- Executing Hangup("Zap/4-1", "") in new
stack== Spawn extension (nec_pri, 2688, 3) exited non-zero on
'Zap/4-1' -- Hungup
'Zap/4-1'MDBRIDGE*CLIThank
you,--LJ
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Re: [Asterisk-Users] Hey admin: Do we have to have a 92-char reply-to header?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 27 August 2004 12:32 am, Brian Capouch wrote: I don't know who else may be suffering from this, but the ultra-long Reply-to: header seems to break my mail reader. I have been suffering the zanies for the last week or so--mainly showing up as the scrollbar disappearing off the right side of my mail window. Tonight I figured out that it's due to the browser reacting to fit the length of the header. The fix was to stretch my mail window out to about 24, occupying my whole screen. This is Mozilla 1.7/Linux, Slackware 9.0. Thanks. B. This is a Mozilla bug. If you can report it to Mozilla. I use 21 screens running at 1600x1200, the point being that I could not imagine NOT using the whole screen for my email client. I want the one-view see-it-all, view. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBL2AbljK16xgETzkRAsPkAKCIzJYnrJplEMpca8FFt7ecMdpKkQCfXmA4 Byz9F8Pj12UgO8jYCUXfU7Y= =o+3I -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe()
-Original Message- From: Larry Shields [mailto:[EMAIL PROTECTED] Sent: Friday, August 27, 2004 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() If I assign the DID to ring extension SIP/2000 and then after time-out send it to MeetMe() or Playback() it works and the caller hears the .gsm file. Any assistance in solving this problem is appreciated. [nec_pri] ; Digital PRI from the NEAX2400 exten = 2688,1,Wait,3 exten = 2688,2,MeetMe,|Mps exten = 2688,3,Hangup I had a similar problem with my system, and I was able to fix the problem by executing Answer before I entered any other applications. Using your previous example: exten = 2688,1,Answer exten = 2688,2,Wait,3 exten = 2688,3,MeetMe,|Mps exten = 2688,4,Hangup Hope this helps, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXOs
I personally am running a couple of X100p cards, a couple of SPA-3000s and a T100P. The X100p cards seem mostly flawless, I do have issues if I am trying to use both at the same time. I suspect it's due to interrupt sharing, it just hasn't bothered me enough to go fix it. The other issue with the X100p (which isn't a big deal eaither..) is an echo when I start the call. I can hear my own echo for 5-10 seconds then it goes away and I don't hear it any more. (This is SIP-*-X100p-PSTN.) I suspect it's the echo training learning?? The SPA-3000 seems to work well also with little or no echo. I put it in-line with my phone at work. So, all my work phone calls go in/out on this with no real problems. The only issue that I have (and again, it hasn't bothered me enough to go fix it yet..) is two stage dialing. When I place an outbound call, * gives me dial tone. I dial and the line starts ringing. The SPA-3000 picks up, I hear dial tone, it dials the number and rings. Although, I guess it is nice to know what's going on with the call. I'm also using another SPA-3000 at my parents house. (They run a bed and breakfast - so 911 is an issue there.) I route all outbound calls to 911 through the FXO port. Other than that, the FXO port isn't really used. (I'm trying to talk them into letting it take their voicemail.. but they're not quite ready for that.) Then I route the bat phone to my *. The T100P is mostly in a testing phase right now. We have a test PRI at the office (used for testing dial up internet equipment and our Interactive Intelligence phone system at the office.) I have that run into a T100P in a * box. (400Mhz Celeron right now.. ) I route all my outbound local calls through that box. I would like to use VOIP for outbound long distance at the office. However, due to internal management issues we can't directly connect the Interactive Intelligence box directly to VOIP. So, I was going to use pri_net signaling with the T100P and feed an ISDN PRI into the II box. All of these are connected to one of 4 different * boxes through IAX2. ie.. I dial out from the house, it goes down to a * box at the office then over to the dev box with the T100P. (Office drops out through then IAX2 transfer.) All that seems to work very well.. with little or now problems. (* has a very high backgroundability factor.. ie, put it in place, forget it, and just use it.) Tom On Aug 27, 2004, at 10:41 AM, [EMAIL PROTECTED] wrote: Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least to me) although several in the user community make refernce to new firmware that might address the issue, real soon now. I see a lot of activity recently on-list about the TDM-400. Of course, mentions on-list are more than likely the result of people having problems. We don't hear about people who have no issues with a product. So, the nature of my inquiry is to explore how many people out here have good/great experiences with the various small FXO adapters? While the TDM-400 is my next possible purchase I'd also like to hear about devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so many products being offered I would hope that we have some collective experience with each one. Thanks, Michael Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713)861-4005 o(800)905-6412 f(713)864-8668 c(713)201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXOs
I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least to me) although several in the user community make refernce to new firmware that might address the issue, real soon now. I see a lot of activity recently on-list about the TDM-400. Of course, mentions on-list are more than likely the result of people having problems. We don't hear about people who have no issues with a product. So, the nature of my inquiry is to explore how many people out here have good/great experiences with the various small FXO adapters? While the TDM-400 is my next possible purchase I'd also like to hear about devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so many products being offered I would hope that we have some collective experience with each one. Have ran both the x100p's and tdm04b (4-port fxo) for several months. In the US, both are usable however the software drivers still are in need of some serious help. Most of the issues revolve around: - interrupts, pci bus or something like that resulting in echo (which appears to be system/motherboard dependent), and echo cancellation functions that have only a limited range, - transmission levels (rxgain/txgain) are not anywhere near useful outside a rather limited range (without causing other problems) - less then stable causing failures that result in having to reboot the entire system (stopping and restarting * and the drivers do not always work). - callerid reliability runs about 80% of what a cheap analog phone bridged onto the pstn line displays. I don't know of anyone that is actually working on correcting the issues, and part of that is likely the result of most users inability to document the problems (since its oftentimes necessary to get the interfaces back up as soon as possible). I've tried the Mediatrix 1204 gateway, and although it works, support is less then acceptable for any serious production use. Echo was non- existant, but lots of other issues when attempting to use it with anything other then four exactly-equal pots lines. The company appeared to be near bankrupcy, therefore any/all software upgrades are chargable. Support is limited to their resellers only and a large percentage of those are not familiar with any voip systems other then whatever commercial products they happen to sell. The company's focus is still in the toll bypass arena using the 1104 and 1204 together. I've not tried most of the other 4-port (or so) boxes as they seem to be very over-priced. I've got a spa3000, but have not had enough time to mess with it in any serious manner. The small amount of testing that I've done make it appear to be a nice box. Hope to learn more about it in the next few weeks. The market seems to be lacking a reasonable/reliable product in the two-to-six fxo port range. Above six ports, seems the channel bank approach becomes more economical, reliable, and easier to support. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Someone please try MeetMe MOH with latest CVS and GS phone
I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem, although I don't know a) whether it is specific to GS phones, and b) whether Mark has a GS phone to try. Please could others with a similar setup try it? The app command is MeetMe with the M flag. The bug report is at http://bugs.digium.com/bug_view_page.php?bug_id=0002312 Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe()
You should be able to hear the audio - a sound card is not involved. Try inserting an answer command in the dialplan before you try to play something. Like Answer Wait (if you want) Playback Hangup Should work (using the proper dialplan commands) Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Shields Sent: Friday, August 27, 2004 9:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a single DID number that rings in from the NEC IPX on PRI Span 1, trunk group 1. If I assign the inbound DID to ring an extension on Asterisk, ie. SIP/2000, it in fact rings and when answered I have a complete 2-way voice path. If I change the destination of the inbound DID from SIP/2000 to MeetMe() or Playback(), Asterisk will answer and I can see from the CLI the .gsm file being played but there is no playback audio heard on the calling extension. If I assign the DID to ring extension SIP/2000 and then after time-out send it to MeetMe() or Playback() it works and the caller hears the .gsm file. Any assistance in solving this problem is appreciated. What follows are two examples from what I tried in extensions.conf: This works but is not desirable: [nec_pri] ; Digital PRI from the NEAX2400 exten = 2688,1,Wait,1 exten = 2688,2,Dial(SIP/2000,3,Tr) exten = 2688,3,Wait,1 exten = 2688,4,MeetMe,|Mps exten = 2688,5,Hangup This will answer, but there is no audible playback on the channel: [nec_pri] ; Digital PRI from the NEAX2400 exten = 2688,1,Wait,3 exten = 2688,2,MeetMe,|Mps exten = 2688,3,Hangup This is what is displayed from the CLI while the calling station is connected via PRI: -- Accepting call from '2502' to '2688' on channel 0/4, span 1 -- Executing Wait(Zap/4-1, 3) in new stack -- Executing MeetMe(Zap/4-1, |Mps) in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (nec_pri, 2688, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' MDBRIDGE*CLI Thank you, --LJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Max TNTs
Hello, Im having some problems with audio on outbound sip to pstn calls from Asterisk to a Max TNT. When I place the calls it connects but I hear pulsing / clicking for the first second or two of the call. My service provider seems to think that the issue may be a result of improper handling of the RTP extension headers. Has anyone else experienced anything like this? Thanks, Ken --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 8/24/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
OK. You need one of the following: Home Highway Business Highway ISDN2e I can confirm that * works happily with all three - my office lines are (for various reasons, none of which apply any more!) on Business Highway. Heh, good old BT. I've never tested voice over Business Highway, as every BT engineer/support/sales person I've spoken to swore blind that it wouldn't work - and in BT's eyes, if they say it won't work, it's unsupported, therefore, if it breaks - you're on your own. Also, I don't believe you can get the full range of 'BT Select Services' or whatever they call them today on the Highway lines (things like Call Deflection, and even caller id on the home highway lines, I believe) BT fully support voice on Business Highway, they just assume that most people will use it for data. Most of BT's 'Digital Select Services' are not available on Home Highway, but are available on Business Highway. As I recall the only service not available on Business Highway, but is available on ISDN 2e is DDI across multiple 2B lines. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
Well I've just called BT, the confirmed to me that MSNs can only work with PTMP and DDIs with PTP. As for the seqential MSN issue, they have Complete tosh! As I said earlier, we've got it - and have ordered it with additional lines as well, if you really want it, just argue more, and talk to a specialist if required. However, Is there any reason (other than cost) to use MSNs over DDIs or the other way round? There is no real reason, but with ISDN2e you will be able to spread the DDI across multiple lines, something they won't do with MSN. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] xlite Problems
- Original Message - From: Tim Jackson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 27, 2004 7:55 AM Subject: [Asterisk-Users] xlite Problems -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and asterisk crashes. I'm running the CVS from yesterday. Any ideas? In XTEN you need to turn off silence suppresion. AdvacedAudio SettingsSilence SettingsTransmit Silence = YES Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
