[asterisk-users] PRI dropouts - solution I hope...
Kevin Withnall
kevin at ilb.com.au
Tue Jul 18 19:42:29 MST 2006
our setup is asterisk only. (trixbox 1.1.1)
I put the pri debug span command in and then generated some logs.
I saved a selection of a working connection and a failed connection and
ran diff over them.
Heres the output if anyone wants to have a look....
Now, obviously this is long but theres a definate difference in pri
signalling after the "dialparties.agi" work.
2c2
< < Call Ref: len= 2 (reference 207/0xCF) (Originator)
---
> < Call Ref: len= 2 (reference 209/0xD1) (Originator)
10c10
< < [18 03 a1 83 81]
---
> < [18 03 a1 83 83]
22c22
< -- Making new call for cr 207
---
> -- Making new call for cr 209
31c31
< > Call Ref: len= 2 (reference 207/0xCF) (Terminator)
---
> > Call Ref: len= 2 (reference 209/0xD1) (Terminator)
33c33
< > [18 03 a9 83 81]
---
> > [18 03 a9 83 83]
38c38
< -- Accepting call from '412453846' to '42270001' on channel 0/1,
span 1
---
> -- Accepting call from '412453846' to '42270001' on channel 0/3,
> span 1
61c61
< -- Executing AGI("Zap/1-1",
"recordingcheck|20060719-122847|1153276127.334") in new stack
---
> -- Executing AGI("Zap/1-1",
> "recordingcheck|20060719-123001|1153276201.338") in new stack
63c63
< recordingcheck|20060719-122847|1153276127.334: Inbound recording not
enabled
---
> recordingcheck|20060719-123001|1153276201.338: Inbound recording not
> enabled
87,108c87,88
< > Protocol Discriminator: Q.931 (8) len=9 < > Call Ref: len= 2
(reference 207/0xCF) (Terminator) < > Message type: ALERTING (1) < > [1e
02 81 88]
< > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Private network serving the local user (1)
< > Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
< -- SIP/601-e7e4 is ringing
< -- SIP/601-e7e4 answered Zap/1-1
< > Protocol Discriminator: Q.931 (8) len=14 < > Call Ref: len= 2
(reference 207/0xCF) (Terminator) < > Message type: CONNECT (7) < > [18
03 a9 83 81] < > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI
Spare: 0, Exclusive Dchan: 0
< > ChanSel: Reserved
< > Ext: 1 Coding: 0 Number Specified Channel
Type: 3
< > Ext: 1 Channel: 1 ]
< > [1e 02 81 82]
< > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Private network serving the local user (1)
< > Ext: 1 Progress Description: Called
equipment is non-ISDN. (2) ]
< < Protocol Discriminator: Q.931 (8) len=5 < < Call Ref: len= 2
(reference 207/0xCF) (Originator) < < Message type: CONNECT ACKNOWLEDGE
(15)
---
> -- SIP/601-4a15 is ringing
> -- SIP/601-4a15 answered Zap/1-1
110,111c90,91
< < Protocol Discriminator: Q.931 (8) len=13 < < Call Ref: len= 2
(reference 207/0xCF) (Originator)
---
> < Protocol Discriminator: Q.931 (8) len=9 < Call Ref: len= 2
> (reference 209/0xD1) (Originator)
116,118d95
< < [1e 02 82 88]
< < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Public network serving the local user (2)
< < Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
120,121c97
< -- Processing IE 30 (cs0, Progress Indicator)
< -- Channel 0/1, span 1 got hangup request
---
> -- Channel 0/3, span 1 got hangup request
124c100
< > Call Ref: len= 2 (reference 207/0xCF) (Terminator)
---
> > Call Ref: len= 2 (reference 209/0xD1) (Terminator)
132c108
< < Call Ref: len= 2 (reference 207/0xCF) (Originator)
---
> < Call Ref: len= 2 (reference 209/0xD1) (Originator)
--
Kevin Withnall
ILB Computing
PH: 02 4227 0001 Mobile: 0412 453 846
FAX: 02 4227 0081
http://kevin.withnall.com/
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James
Sturges
Sent: Monday, 17 July 2006 10:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] PRI dropouts - solution I hope...
Is your dial plan very simple, ie bypass FREEPBX etc, to make sure no
problems.
There are also debug command in the CLI:
pri debug span Enables PRI debugging on a span
pri intense debug span Enables REALLY INTENSE PRI debugging
pri no debug span Disables PRI debugging on a span
pri show debug Displays current PRI debug settings
pri show span Displays PRI Information
maybe also set the debug and verbose and see what it says.
Is your set the same, is Asterisk between the line and the PBX or just
Asterisk?
Have you tried just using Trixbox?
Thanks
James
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin
Withnall
Sent: Monday, 17 July 2006 7:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] PRI dropouts - solution I hope...
My files were almost exactly the same. We only have 10 channels and the
clid signaling was different.
We are however still getting the same problems. I moved the box closer
to the optomux (now we have 2m cable from the optomux to the asterisk
box.)
Any other ideas? We still are having the same problems and also, some
dropouts in the middle of calls.
Could the card be faulty ? I purchased it from ebay second hand.
PS. What does the "Transfer=yes" do ?
Thanks.
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James
Sturges
Sent: Saturday, 15 July 2006 10:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] PRI dropouts - solution I hope...
Hi, had a few ask for this so thought may be of interest to the list.
This is actually for the following setup:
Telstra ISDN30 <---------> Asterisk <---------> BP250 PABX
The ISDN10, 20, 30's are all the same physical link, but you may need to
change the bchan and dchan settings for ISDN 10 or 20.
We have had lot of issues over 12 months, including physical cable
issues, etc. But this config has passed Telstra test equipment both on
site and in the exchange. The calls dropping out (for us) are timing
issues do to telling Asterisk to gets it synch from the Telstra line and
providing synch to the PABX.
Anyway, Here it is, does not look like much but have had experts working
on it for a while.
The system handles 1800 - 2000 calls per day.
Thanks
James
ZAPATA.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
context=te405p-intelstra
pridialplan=local
signalling=pri_cpe
callerid=asreceived
channel=>1-15, 17-31
group=4
context=te405p-frombp250
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=>94-108, 110-124
ZAPTEL.conf
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124
loadzone=au
defaultzone=au
________________________________
From: James Sturges [mailto:thinking at 1am.com.au]
Sent: Saturday, 15 July 2006 12:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] PRI dropouts
Have had L O T S of trouble like this, the settings zap config files
seem to have to e exact, please send email to thinking at 1am.com.au and I
will send config files.
Thanks
James
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin
Withnall
Sent: Saturday, 15 July 2006 11:05 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] PRI dropouts
Recently we cut over to using asterisk (trixbox 1.1.1) for our
production system.
We are using a TE110P digium card (Primary rate) with a Telstra onramp
10.
Sometimes when people call, on their end it doesn't seem to connect. On
our end, we get caller id, it passes ok to the sip phone but then no-one
is there.
Anyone have any similar problems and worked out how to solve it ?
Thanks.
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