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tzafrir.cohen at xorco...
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 Posted: Thu Jan 10, 2008 7:04 pm Post subject: [asterisk-users] HFC-S zap channels always busy |
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On Fri, Jan 11, 2008 at 12:45:48AM +0100, Jaap Winius wrote:
| Quote: | Quoting Tzafrir Cohen <tzafrir.cohen at xorcom.com>:
| Quote: | What do you mean by "In Use"?
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# cat /proc/zaptel/*
Span 1: ZTHFC1 "HFC-S PCI A Zaptel Driver card 0 [TE]" (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
| Quote: | When Asterisk runs, they are "(In use)" - by Asterisk.
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I would hope so. I do see:
# asterisk -rx 'pri show spans'
PRI span 1/0: Provisioned, Up, Active
and
# asterisk -rx 'zap show channels'
Chan Extension Context Language MOH Interpret
pseudo isdn-in en default
1 isdn-in en default
2 isdn-in en default
This is when I use the vzaphfc modules. However, any attempt to dial
out via ISDN results in errors like:
-- Executing [0654314647 at phones:1] Dial("SIP/1000-081f3220",
"Zap/g0/0654314647 at channels||r") in new stack
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'@channels'?
What is the Dial command you use?
Can you post the relevant part of your diaplan?
Also, try:
bri debug span 1
And see if you see any traffic there when you try to call out.
In addition: are you sure that there are channels set for group=0 ?
Maybe try a channel directly: Zap/1 or Zap/2 instead of Zap/g0 .
| Quote: | [Jan 11 00:12:15] WARNING[1354]: app_dial.c:1130 dial_exec_full:
Unable to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1000-081f3220' status is 'CONGESTION'
[Jan 11 00:12:15] NOTICE[1354]: cdr.c:434 ast_cdr_free: CDR on channel
'SIP/1000-081f3220' not posted
In this case, the rule that I use in extensions.conf for dialing out is:
exten => _X.,1,Dial(Zap/g0/${EXTEN}@channels,,r)
| Quote: | signalling=bri_cpe
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I've tried that, but unless I'm doing something else wrong
(hopefully), using "signalling=bri_cpe" instead of
"signalling=bri_cpe_ptmp" makes no difference.
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--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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jwinius at umrk.to
Guest
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| Posted: Fri Jan 11, 2008 9:53 am Post subject: [asterisk-users] HFC-S zap channels always busy |
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Quoting Tzafrir Cohen <tzafrir.cohen at xorcom.com>:
| Quote: | What is the Dial command you use?
Can you post the relevant part of your diaplan?
|
exten => _X.,1,Dial(Zap/g0/${EXTEN}@channels,,r)
| Quote: | In addition: are you sure that there are channels set for group=0 ?
Maybe try a channel directly: Zap/1 or Zap/2 instead of Zap/g0 .
|
Aha! So, that's what /g*/ refers to: the group number specified in
zatapa.conf! I set this to g0 after blindly following your advice of
27 December last year:
"So basically dial to Zap/g0/NUMBER and it should dial it to
and it should dial it to your provider."
The problem was that I had set "group=1" in the zatapa.conf. Now that
I've changed the dial command to Zap/g1/, the situation is vastly
improved:
== Primary D-Channel on span 1 up
and
-- Executing [0654314647 at phones:1] Dial("SIP/1000-081f68b0",
"Zap/g1/0654314647 at channels||r") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0654314647 at channels
-- Channel 0/1, span 1 got hangup request, cause 28
-- Hungup 'Zap/1-1'
This is when I try to make a call, which looks much better now.
In conclusion, there was nothing wrong with the channels being
described as "in use" in /proc/zaptel/*. However, the "Unable to
create channel of type 'Zap' (cause 34 - Circuit/channel congestion)"
error messages that I was getting as a result using the wrong group
number in the command to dial out were misleading. If only Asterisk
had been able to distinguish between congestion and a group that
simply didn't exist.
Of course, I still have a few other problems, but they're different
and seem minor in comparison, so I won't mention them in this thread.
This particular problem, however, is hereby solved.
Many thanks, Tzafrir!
Cheers,
Jaap |
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tzafrir.cohen at xorco...
Guest
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| Posted: Fri Jan 11, 2008 3:26 pm Post subject: [asterisk-users] HFC-S zap channels always busy |
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On Fri, Jan 11, 2008 at 03:53:11PM +0100, Jaap Winius wrote:
| Quote: | Quoting Tzafrir Cohen <tzafrir.cohen at xorcom.com>:
| Quote: | What is the Dial command you use?
Can you post the relevant part of your diaplan?
|
exten => _X.,1,Dial(Zap/g0/${EXTEN}@channels,,r)
| Quote: | In addition: are you sure that there are channels set for group=0 ?
Maybe try a channel directly: Zap/1 or Zap/2 instead of Zap/g0 .
|
Aha! So, that's what /g*/ refers to: the group number specified in
zatapa.conf! I set this to g0 after blindly following your advice of
27 December last year:
"So basically dial to Zap/g0/NUMBER and it should dial it to
and it should dial it to your provider."
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(This is because with genzaptelconf and zapconf I put all "trunks" also
in group 0)
| Quote: |
The problem was that I had set "group=1" in the zatapa.conf. Now that
I've changed the dial command to Zap/g1/, the situation is vastly
improved:
== Primary D-Channel on span 1 up
and
-- Executing [0654314647 at phones:1] Dial("SIP/1000-081f68b0",
"Zap/g1/0654314647 at channels||r") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0654314647 at channels
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Again, you're calling an incorrect number. You dial to a number that
includes the string "@channels". This is of course a number your telco
does not know how to handle. Also you may need to add
"pridialplan = unknown" in zapata.conf.
| Quote: | -- Channel 0/1, span 1 got hangup request, cause 28
-- Hungup 'Zap/1-1'
This is when I try to make a call, which looks much better now.
|
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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jwinius at umrk.to
Guest
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| Posted: Fri Jan 11, 2008 5:32 pm Post subject: [asterisk-users] HFC-S zap channels always busy |
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Quoting Tzafrir Cohen <tzafrir.cohen at xorcom.com>:
| Quote: | | Quote: | -- Executing [0654314647 at phones:1] Dial("SIP/1000-081f68b0",
"Zap/g1/0654314647 at channels||r") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0654314647 at channels
|
Again, you're calling an incorrect number. You dial to a number that
includes the string "@channels". This is of course a number your telco
does not know how to handle. Also you may need to add
"pridialplan = unknown" in zapata.conf.
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At first I didn't understand what you meant by "'@channels'?" in your
last response. Now I see. It was another example of me using someone
else's config without knowing what it did. Now I can dial out via ISDN
as well as receive calls, so there's no longer any need to start up
any new threads either. At least, not immediately. 
Thanks a million, Tzafrir!
Cheers,
Jaap |
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