Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing '8801234' [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:2030 zt_call: Deferring dialing... -- Called 3/8801234 [Feb 25 02:37:00] WARNING[7194]: chan_zap.c:3835 zt_handle_event: Detected alarm on channel 3: No Alarm -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) [Feb 25 02:37:00] NOTICE[7082]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 3 So the call fails and if I weren't using a test extension: exten => 2111,1,Dial(Zap/3/8801234) it would proceed in the dialplan. asterisk]# cat /proc/zaptel/1 Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 Where do I go with this? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20080225/3b048c10/attachment.pgp
Johan Sandgren
2008-Feb-25 09:34 UTC
[asterisk-users] Detect DTMF of 24 channels simultaniously?
Hi! I have 3 TDM800P with 8 incoming lines each. Is it possible to detect DTMF-tones simultaniously on all 24 Zap-channels? Or is there a limitation on this? I guess dtmf-detection is a kind of DSP-operation, which consumes a lot of CPU-power. Will the zap-driver be able to "sample" all channels at the same time, and also detect DTMF-tones? Anyone knows? Sincerely, Johan
On Mon, Feb 25, 2008 at 3:42 AM, Anthony Messina <amessina at messinet.com> wrote:> Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing > out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. > I get the following: > > -- Starting simple switch on 'Zap/1-1' > -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack > [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing '8801234' > [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:2030 zt_call: Deferring dialing... > -- Called 3/8801234 > [Feb 25 02:37:00] WARNING[7194]: chan_zap.c:3835 zt_handle_event: Detected > alarm on channel 3: No Alarm > -- Hungup 'Zap/3-1' > == Everyone is busy/congested at this time (1:0/0/1) > [Feb 25 02:37:00] NOTICE[7082]: chan_zap.c:6678 handle_init_event: Alarm > cleared on channel 3 > > So the call fails and if I weren't using a test extension: > exten => 2111,1,Dial(Zap/3/8801234) > > it would proceed in the dialplan. > > asterisk]# cat /proc/zaptel/1 > Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) > > 1 WCTDM/0/0 FXOKS (In use) > 2 WCTDM/0/1 > 3 WCTDM/0/2 FXSKS (In use) > 4 WCTDM/0/3 > > > Where do I go with this? > > -- > Anthony - http://messinet.com - http://messinet.com/~amessina/gallery > 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >Look at http://bugs.digium.com/view.php?id=11855. sean
Anthony Messina wrote:> Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing > out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. > I get the following:This should be fixed in Zaptel 1.4.9.2. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc.
Is this only on the "_64" zaptel or will affect ALL zpatel 1.4.9 ? -----Original Message----->From: Russell Bryant <russell at digium.com> >Sent: Feb 28, 2008 6:11 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> >Subject: Re: [asterisk-users] TDM400P dialout problem > >Anthony Messina wrote: >> Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing >> out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. >> I get the following: > >This should be fixed in Zaptel 1.4.9.2. > >-- >Russell Bryant >Senior Software Engineer >Open Source Team Lead >Digium, Inc. > >_______________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users