[asterisk-users] Linksys/Sipura 3K, Calls Timing Out
Hi All, I have a Linksys SPA-3000 [Hardware version 3.0.0(1178), Software version 3.1.10(GWd)], with both the FXO and FXS interfaces registering with asterisk via SIP seperatley. I also have a Cisco 7940 and 7960 using the sccp2 (chan_sccp) driver, and a couple of IAX softphones Both inbound and outbound calls to/from the FXO interface time out after around 17-20 minutes. With SIP debug turned on, it looks like the call was just ended "normally". This problem doesnt occur with IAX<->SCCP calls, just those via the SPA-3K FXO interface. Ive checked all the timeouts in the Linksys configuration, and set them all way higher than 17-20 minutes. I've tried with both G729, and uLaw CODECs, same thing. I've tried turning off silence detection, and the hangup detection is set correctly for the UK. Has anyone else had this happen, or any idea what the problem might be ? As you might imagine, its rather frustraiting to be half way thru a call, and it just hang up on you! All the Best Iain ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting outbound trunk
On Tue, Aug 29, 2006 at 02:18:32PM +1000, Devraj Mukherjee wrote:
> The simplest way I can think of solving this is using prefixes, so
> someone appends a 0 or 1 and the dialplan puts the call through the
> selected trunk, where 0 being voip and 1 being PSTN.
Whats wrong with something like this :
exten => _91X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _92X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _93X.,1,Dial(IAX2/iaxprov/${EXTEN:2})
Users would dial 91 to dial outbound on SIP Provider 1, 92 for
outbound on SIP Provider 2, and 93 for outbound on IAX. Personally
I use 9X for automatic routing (along with some sane forced routing,
ie local, emerg calls etc), and am planning on using 8X for manual
forced routing.
> I have figured out how to use a Substring like function to extract the
> number out of the dialed extension. My question is how do I make a
> decision in the dialplan to dynamically select a trunk for the call?
> Is there a SetIf function or an If function by itself?
Checkout the command GotoIf()
Heres an example that I use to in my exten Macro, that does
slightly different things depending on the number range the
extension dialed is from:
[macro-exten]
exten => s,1,GotoIf($[${ARG1:0:1} = 1]?11:21) ; Did we call a real ext ?
exten => s,11,SetVar(TODIAL=${ARG2}/${ARG1}); Yes so we have the ext
exten => s,12,Goto(91) ; Jump to Dial() routint
exten => s,21,GotoIf($[${ARG1:0:2} = 20]?31:41) ; Did we call a virt or soft ?
exten => s,31,SetVar(VMBOX=${ARG1}) ; Virt, So vm is the same
exten => s,32,SetVar(TODIAL=${VIRT[${ARG1}]}) ; Grab the list of real exts
exten => s,33,Goto(91) ; Jump to the dial routine
exten => s,41,SetVar(VMBOX=20${ARG1:1:1}) ; Soft, So vm is the virt
exten => s,42,SetVar(TODIAL=${ARG2}/${ARG1}); But it is a real ext
exten => s,43,Goto(91)
exten => s,91,Dial(${TODIAL},25,Tt)
exten => s,92,GotoIf($[${ARG1:0:1} = 2]?93:94) ; Do we need to handle vm ?
exten => s,93,GoSub(s-${DIALSTATUS},1)
exten => s,94,Hangup()
exten => s-NOANSWER,1,Voicemail(u${VMBOX}) ; Virtual extensions have
exten => s-BUSY,1,Voicemail(b${VMBOX}) ; VM, so transfer caller
exten => s-CHANUNAVAIL,1,Voicemail(u${VMBOX}) ; Offline, so transfer call
I have a dialplan where 1xx are real extensions, with no voicemail,
20x are virtual extensions, identified with an induvidual, with voicemail, and
2xy are extensions assoiated with the same induvidual as the virtual
number (ie 21x are all linked to 201 etc..)
HTH
Iain
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Re: [Asterisk-Users] mini itx
On Thu, Jun 23, 2005 at 11:39:21AM -0500, jltaylor wrote: > I've seen the embedded posts. > Is anyone running Asterisk on the MINI ITX? Yes, no problems, I have an X100P in the PCI slot, but its only a single POTS line. I used the MII board, but only because thats what I had avaliable. Iain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 with VIA Chipset
On Wed, May 11, 2005 at 11:15:29PM +0200, Armin Lediger wrote: > I am trying to install asterisk 1.0.7 on a VIA EPIA 5000 board - anyone > of you already managed to do so? I got V1.0.6 running, but 1.0.7 seems > not to compile. I have the MII-12000 board, and the debian packages work for me, although to be fair, its only the binary packages I've tried, never tried to compile the source packages. I'm not aware of anything (that I can recall) with the VIA EPIA boards that would prevent most userspace stuff from working (bar some mplayer bits) Best Regards Iain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outgoing calls, X100P
Hi Mehmet, On Tue, May 03, 2005 at 11:20:44AM -0400, You wrote: > I tried that and it didn't work. Then I decided to use a different phone > line. I had not thought about this before, it just didn't occur to me. > And everything worked fine. The phone line that doesn't work is my ADSL > line. Hmm. interesting that my line was also an ADSL line. Did you just try the values for rx and txgain that I gave you, or did you go higher ? I started at 10db gain (Horrible echo, but dialed), then I worked down, until I found the lowest gain that gave me minimal echo, but still worked. > Wall to splitter, one side going to ADSL router the other going > into a fax machine and than from fax machine going into X100P. I > remember seeing a post about this before. I'll have to check into that. I'd suggest trying it Wall -> Splitter -> X100P, and see what that does. Asterisk can act as a fax machine anyway. All the Best Iain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outgoing calls, X100P
On Mon, May 02, 2005 at 01:02:34PM -0400, Mehmet Tolga Avcioglu wrote: > I can't seem to be able to make outgoing calls with X100P card. I can > receive calls fine and it picks up the line and sends the tones, but the > telco doesn't recognize them. While the tones are sent I continue to > hear the dial tone on the line when I pick up a parallel. I also cannot > dial from the parallel until X100P hangs up the line. I had the same trouble, here in the UK. What do you have rxgain and txgain set at in zapata.conf ? I found I had to raise the txgain to 0.25 to get it to work on my British Telecom line (Cable Provider's line was fine) I also had to up even further (to 3.0) in order to get inbound fax to work, although I think I can back that off a little. Of course, increasing the gain also gives me an echo issue, in proportion to the gain values, so I suspect I have an impedance issue. YMMV of course. Hope that Helps Iain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] UK PSTN Calling From OH323 Problem
Hi Cameron,
You Wrote:
On Thu, Apr 14, 2005 at 08:47:38AM +1200, Cameron Beattie wrote:
> Couple of things I noticed:
> Dunno why you'd have s extension in [voip-h323] context.
> Format of Dial command seems wrong
> Try putting the specific number you dial in extensions.conf e.g.
> exten => 020,1,Dial(Zap/1/020)
> exten => _9X.,1,Dial(Zap/1/${EXTEN:1})
Thanks for the hints. Turned out I needed to up the txgain to 0.25,
in order to get the DTMF tones to be picked up at the exchange end.
Spoke with a colleague, who mentioned that 1,4, and 7" are the lowest
frequency DTMF tones, so maybe that's related to "1471" working, and
not other numbers.
Best Regards
Iain
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[Asterisk-Users] UK PSTN Calling From OH323 Problem
Hi All,
I'm just starting with Asterisk, so this may be something very
simple. I'm using a X100P, and a softphone (GnomeMeeting) with
OH323 providing the linkage into Asterisk. The (very) simple
setup looks like this:
PSTN---X100PAsterisk---OH323 GateKeeper---GnomeMeeting
(zap channel) (OH323 Channel)
I'm slowly working thru my own Dial Plan, bit by bit. I finally
got my head around the concepts and theory of extensions.conf,
but found the supplied file rather complex and confusing.
So, in order to test my understanding, and actually get some
experience of writing it myself, I started from a clean sheet of
paper.
I've sorted incoming calls from the POTS line, so they get
sent to the softphone. But I'm having difficulty getting
outbound calls to the PSTN line from the softphone to work
correctly.
extensions.conf contains the following:
---BEGIN INCLUDE FILE---
[general]
static=no
writeprotect=yes
[global]
RINGTIME=>3
POTSOUT=>Zap/1
[default]
exten => s,1,Wait,1
exten => s,2,Answer ; Answer The Incoming Call
exten => s,3,Dial,OH323/iain; Transfer to the SoftPhone
exten => s,4,HangUp ; Hang Up
[voip-h323]
exten => s,1,Wait,1
exten => s,2,Answer ; Answer The Incoming Call
exten => s,3,DigitTimeout,15; Give us time to enter the real num
exten => _9X.,1,Dial,Zap/1/${EXTEN:1} ; Try This
---END INCLUDE FILE---
So, What I was hoping this would do, is dialing 9 followed by a
number, would send the call out the POTS line. Well it sorta works.
If I dial (from the softphone) 91471 (1471 is the number you call in
the UK to see who called your POTS line last), it works (seems to take
an age to dial, and connect, but does work)
However, if I call other numbers, eg my own mobile, or another PSTN
number, then although a 'show channels' shows the correct number, it
never seems to reach the number that (I think) it dialed
Asterisk version is Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k from Debian
Sid.
Anyone have any suggestions as to what I'm doing wrong ? Is there
anything specific to the UK ? Do I need to add something to ensure
it knows how to dial in the UK ?
All the Best
Iain
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