[asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

2008-09-24 Thread Nathan Dennis 
   Timestamp: 09855ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
-- Hungup 'IAX2/brisbane-16384'
  == Spawn extension (iax2brisbaneout, 5510, 1) exited non-zero on
'SIP/1406-b7b2b530'
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/1406-b7b2b530,
IAX2/brisbane/[EMAIL PROTECTED]) in new stack
-- Called brisbane/[EMAIL PROTECTED]
-- Hungup 'IAX2/brisbane-16385'
Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass:
HANGUP
   Timestamp: 16939ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
   CAUSE CODE  : 16
 
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
   Timestamp: 3ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   VERSION : 2
   CALLED NUMBER   : h
   CODEC_PREFS : (g729)
   CALLING NUMBER  : 1406
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME: Nathan Dennis
   LANGUAGE: en
   CALLED CONTEXT  : internal
   USERNAME: cairns
   FORMAT  : 256
   CAPABILITY  : 57600
   ADSICPE : 2
   DATE TIME   : 2008-09-24  18:35:44
 
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
   Timestamp: 6ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   CAUSE CODE  : 0
 
  == Spawn extension (iax2brisbaneout, h, 1) exited non-zero on
'SIP/1406-b7b2b530'
 Extension Changed 1406[internalhints] new state Idle for Notify User
1401 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1419 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1415 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1402 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1404 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1411 (queued)
 Extension Changed 1406[internalhints] new state Idle for Notify User
1408 (queued)
-- Incoming call: Got SIP response 500 Internal Server Error back
from 10.10.11.193
Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 006 Type: IAX Subclass:
ACK
   Timestamp: 16939ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
   Timestamp: 3ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
   Timestamp: 00011ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
   AUTHMETHODS : 3
   CHALLENGE   : 842177371
   USERNAME: cairns
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00116ms  SCall: 16385  DCall: 08602 [10.10.51.22:4569]
   MD5 RESULT  : b9f5616e47f5f8e5605868b717d510aa
 
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
INVAL
   Timestamp: 0ms  SCall: 0  DCall: 16385 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
VNAK
   Timestamp: 00118ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
   Timestamp: 6ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   CAUSE CODE  : 0
 
Tx-Frame Retry[001] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00116ms  SCall: 16385  DCall: 08602 [10.10.51.22:4569]
   MD5 RESULT  : b9f5616e47f5f8e5605868b717d510aa
 
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
INVAL
   Timestamp: 0ms  SCall: 0  DCall: 16385 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
VNAK
   Timestamp: 00232ms  SCall: 08602  DCall: 16385 [10.10.51.22:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP
   Timestamp: 6ms  SCall: 16385  DCall: 0 [10.10.51.22:4569]
   CAUSE CODE  : 0

 
 
IAX Debug on second Server
 
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 00091ms  SCall: 06840  DCall: 16384 [10.10.11.22:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: VOICE   Subclass:
136
   Timestamp: 00100ms  SCall: 06840  DCall: 16384 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
   Timestamp: 00088ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
ACK
   Timestamp: 00091ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass:
ACK
   Timestamp: 00100ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 004 Type: VOICE   Subclass:
136
   Timestamp: 00288ms  SCall: 16384  DCall: 06840 [10.10.11.22:4569]
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass:
ACK
   Timestamp: 00288ms  SCall: 06840  DCall: 16384 [10.10.11.22:4569]
Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass

Re: [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

2008-09-24 Thread Nathan Dennis 
Thanks for pointing that out Tony, Should have included that in my first
post.
Below is the version and the IAX config for each end

Server 1
Version : 1.4.18

IAX2.conf peer details

[brisbane]
type=friend
host=XXX.XXX.XXX.XXX
trunk=yes
context=internal
context=parkinglot
qualify=1
username=XXX
secret=
disallow=all
allow=g729

Server 2 
Version : 1.4.21.2

IAX2.conf peer details

[cairns]
type=friend
host=XXX.XXX.XXX.XXX
trunk=yes
context=internal
context=callagents
context=parkinglot
qualify=1
username=XXX
secret=
disallow=all
allow=g729



Nathan Dennis 
__ 
Integrated Solutions (QLD)P/LPhone: +61 (7) 4044 0300 
Direct: +61 (7) 4044
0302
124 Spence Street   Fax:+61 (7) 4041 6600
CAIRNS QLD 4868Mobile: 0418 608609

Australia 

E-mail: [EMAIL PROTECTED]
Web Site: www.i-solutions.net.au

Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
Sydney
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Wednesday, 24 September 2008 8:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAX Hangup floods link with repeated
VNAKand HANGUP

In article
[EMAIL PROTECTED]
.au,
Nathan Dennis [EMAIL PROTECTED] wrote:
 We have been using asterisk for a while now but have recently needed 
 to install a second server in a remote office and set up a iax trunk 
 between the 2 servers. The dial plan seems to work well when I tested 
 it on the same LAN. However this afternoon I connected the system at 
 the remote office and made some calls. All the calls connect and work 
 fine, voice quality is great no really couldn't have hoped for better.

 Hang up the call and tried to make another call and nothing, the link 
 was not responding, after much trouble shooting I have found that 
 after the call is hung up the 2 asterisk servers seem to go into some 
 kind of loop sending each other message. I have pasted a debug for 
 both servers below that include everything from the start of the call 
 to after hangup. I have cut them short at the VNAK and Hangup cycle 
 just continues for 30seconds or so flooding the link completely.
  
 Any help you may be able to provide would be greatly appreciated

I can't help with your problem, sorry, but anyone who can help will need
to know exactly what version of Asterisk you have at each end.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Xorcom Bri and asterisk crashes

2007-07-09 Thread Nathan Dennis 
Thanks for the input, but I still don't seem to have any luck with the
devices locking up. I've even rebuilt a new system on new hardware and a
new xorcom device but still no good. Once the device locks up that's it
the only way to get zaptel and asterisk back up is to turn them off and
restart the server. The command you have me
 
rmmod xpp_usb; /usr/share/zaptel/xpp_fxloader reset
Works great and will reload the firmware as long as the devices are
frozen. Once they lock up this command will not reload the firmware and
brings up the following errors.

'xpp_fxloader'[16065]: Resetting FPGA Firmware on /dev/bus/usb/005/016
'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (247): bulk_write
failed: error reaping URB: No such device
'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (810): Renumeration to
default failed: errno=-19
'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (203): Releasing
interface: usb: could not release intf 0: No such device
'xpp_fxloader'[16342]: fpga_load failed remoivng with status 237


I'm running Fedora 7, Kernel 2.6.21-1.3194.fc7

Will hopefully be upgrading the kernel tonight if I can get some
downtime to do so.

As for more traces, I can do that, but being reasonably new to this I
will need some help getting them for you.



 
On Thu, Jul 05, 2007 at 09:14:12AM +1000, Nathan Dennis wrote:
 We have recently install an asterisk solution with about 60 physical
 extensions. While the system is running it runs reasonably well (Still
 have a few teething problems) but twice now they have experienced a
 degradation in voice quality and dropped calls and then finally
asterisk
 completely crashes out. Restarting asterisk will work for a little
while
 and it will crash again, each time less time will pass before a crash
 out. The first time I didn't have much logging so I didn't get
anything
 to work with. I have since turned on debugging and following is the
logs
 from the time of the last crash. Can anyone point out where the
problem
 may lay, suggested updates or changes?
  
  
 Jul  4 11:56:51 DEBUG[20042] chan_sip.c: Auto destroying call
 'aca7e8d7fc914018 at 192.168.12.164
http://lists.digium.com/mailman/listinfo/asterisk-users '
 Jul  4 11:56:54 DEBUG[20042] chan_sip.c: Stopping retransmission on
 '6eeb52b53a414a6975facbc22ca10686 at 192.168.10.12
http://lists.digium.com/mailman/listinfo/asterisk-users ' of Request
102: Match
 Found
 Jul  4 11:56:55 VERBOSE[20050] logger.c: -- Accepting voice call
 from '' to '40312688' on channel 0/2, span 5
 Jul  4 11:56:55 DEBUG[20050] chan_zap.c: Enabled echo cancellation on
 channel 14
 Jul  4 11:56:55 DEBUG[20295] pbx.c: Function result is 'CID withheld'
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
 Set(Zap/14-1, CALLERID(name)=Old Main Line:CID withheld) in new
 stack
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
 Goto(Zap/14-1, mainq|q|1) in new stack
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,1)
 Jul  4 11:56:55 DEBUG[20295] pbx.c: Function result is 'false'
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
 Set(Zap/14-1, NightMode=false) in new stack
 Jul  4 11:56:55 DEBUG[20295] pbx.c: Expression result is '0'
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
 GotoIf(Zap/14-1, 0?afterhoursq|q|1) in new stack
 Jul  4 11:56:55 DEBUG[20295] pbx.c: Not taking any branch
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
 GotoIfTime(Zap/14-1, 8:00-17:30|mon-fri|*|*|?businesshours) in new
 stack
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,5)
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
 Set(Zap/14-1, __ALERT_INFO=http://www.example.com
http://www.example.com/ ;info=MainQ) in
 new stack
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Executing
 Queue(Zap/14-1, mainq1|twr|||10) in new stack
 Jul  4 11:56:55 DEBUG[20295] chan_zap.c: Requested indication 3 on
 channel Zap/14-1
 Jul  4 11:56:55 VERBOSE[20295] logger.c: -- Called
 Local/700 at callagents
http://lists.digium.com/mailman/listinfo/asterisk-users 
 Jul  4 11:56:55 VERBOSE[20298] logger.c: -- Executing
 Set(Local/700 at callagents-bc5a
http://lists.digium.com/mailman/listinfo/asterisk-users ,2,
Extension=700) in new stack
 Jul  4 11:56:55 VERBOSE[20298] logger.c: -- Executing
 Set(Local/700 at callagents-bc5a
http://lists.digium.com/mailman/listinfo/asterisk-users ,2,
 __ALERT_INFO=http://www.example.com http://www.example.com/
;info=MainQ) in new stack
 Jul  4 11:56:55 VERBOSE[20298] logger.c: -- Executing
 Dial(Local/700 at callagents-bc5a
http://lists.digium.com/mailman/listinfo/asterisk-users ,2,
SIP/700||tw) in new stack
 Jul  4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on RTP to 524288
 Jul  4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on VRTP to 524288
 Jul  4 11:56:55 DEBUG[20298] chan_sip.c: Outgoing Call for 700
 Jul  4 11:56:55 VERBOSE[20298] logger.c: -- Called 700
 Jul  4 11:56:55 DEBUG[20042] chan_sip.c: (Provisional) Stopping

Re: [asterisk-users] Need Advice/Suggestion

2007-07-05 Thread Nathan Dennis 
Hi Farooq,
  I've done just that for one of our customers. All I did was
add an exten such as *56 that set a custom database value to
nightmode=true. Then as calls come in I just check the database value to
see if it is set to true or not. Note I have asterisk patched with
Bristuff so unless you do as well the hint section will not work.


See Below

exten = *56,hint,DS/56
exten = *56,1,Set(NightMode=${DB(nightmode/active)})
exten = *56,n,playback(service)
exten = *56,n,Gotoif($[${NightMode} = true]?turnoff)
exten = *56,n,Set(DB(nightmode/active)=true)
exten = *56,n,devstate(56,2)
exten = *56,n,playback(activated)
exten = *56,n,hangup()
exten = *56,n(turnoff),Set(DB(nightmode/active)=false)
exten = *56,n,playback(de-activated)
exten = *56,n,devstate(56,0)
exten = *56,n,hangup

Then as a call comes in you just check the value in the database

exten = q,1,Set(NightMode=${DB(nightmode/active)})
exten = q,n,Gotoif($[${NightMode} = true]?afterhoursq,q,1)
exten = q,n,GotoIfTime(8:00-17:30|mon-fri|*|*|?businesshours)


Nathan Dennis 
__ 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Farooq
Ahmed
Sent: Tuesday, 3 July 2007 5:00 PM
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Subject: [asterisk-users] Need Advice/Suggestion

Hi all,
As we know we can configure in astersik like before 5:00pm calls go to
reception and after 5:00 pm calls go to some mobile no. One of my client
requested that he wants to manually shift the dial plan  like above as
he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I
can not give him freepbx  access.
Any idea or solution.
Regards
Farooq
-- 

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[asterisk-users] Problems with SIP Registration on VPN Link

2007-07-04 Thread Nathan Dennis 
Hi,
We are having major problems with a remote site that links to the
head office via a VPN tunnel. The phones will register fine and work for
a few minutes to hours but then will drop their connection and will no
register to asterisk even with a restart of the phone. We have 2 other
remote sites that work exactly same and they are not having any issues
so i believe it has to be be something to do with the network rather
then asterisk but this is the sip debug for a phone trying to register.
Any idea where i should start to look as this has me totally confused as
obviously the phones can communicate with asterisk at all times just
something is causing the registration to get screwed up.
 
Jul  4 09:43:46 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce
received from 'sip:[EMAIL PROTECTED];user=phone'
Transmitting (NAT) to 192.168.12.63:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63
From: Edmonton Boardroom 1
sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805
To: sip:[EMAIL PROTECTED];user=phone;tag=as4d6893cc
Call-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=48f69f92, stale=true
Content-Length: 0
 

---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000
ms
cnsmavs1*CLI
-- SIP read from 192.168.12.63:5060:
REGISTER sip:192.168.10.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15
From: Edmonton Boardroom 1
sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805
To: sip:[EMAIL PROTECTED];user=phone
Contact: sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone
Supported: path
Authorization: Digest username=763, realm=asterisk, algorithm=MD5,
uri=sip:192.168.10.12, nonce=587da437,
response=4bd29b9213057e3e2f3a5270748fbe85
all-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
Expires: 3600
User-Agent: Grandstream GXP2000 1.1.2.23
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,M
ESSAGE
Content-Length: 0
 

--- (14 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.12.63 : 5060 (NAT)
Transmitting (NAT) to 192.168.12.63:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63
From: Edmonton Boardroom 1
sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805
To: sip:[EMAIL PROTECTED];user=phone
Call-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
 

---
Jul  4 09:43:48 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce
received from 'sip:[EMAIL PROTECTED];user=phone'
Transmitting (NAT) to 192.168.12.63:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63
From: Edmonton Boardroom 1
sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805
To: sip:[EMAIL PROTECTED];user=phone;tag=as4d6893cc
Call-ID: [EMAIL PROTECTED]
CSeq: 10005 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=750fc224, stale=true
Content-Length: 0
 

 

Nathan Dennis 


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[asterisk-users] Xorcom Bri and asterisk crashes

2007-07-04 Thread Nathan Dennis 
in 64/64 formats
Jul  4 11:56:59 DEBUG[20295] channel.c: Released clone lock on
'Local/[EMAIL PROTECTED],1ZOMBIE'
Jul  4 11:56:59 DEBUG[20295] channel.c: Done Masquerading
SIP/700-09530a90 (6)
 
The last entry was just before the crash. 
 
We also have this in dmesg (Not sure if its related)
 
NOTICE-xpd_bri: XBUS-01/XPD-10: D-Chan RX Bad checksum: [FE:28=FC] (252)
NOTICE-xpd_bri: XBUS-01/XPD-10: Multibyte Drop: errno=-71
BUG: warning at kernel/softirq.c:138/local_bh_enable() (Not tainted)
 [c042b0cf] local_bh_enable+0x45/0x92
 [c06002bd] cond_resched_softirq+0x2c/0x42
 [c059adf3] release_sock+0x4f/0x9d
 [c05c670d] tcp_sendmsg+0x90b/0x9f9
 [c059adb6] release_sock+0x12/0x9d
 [c05c7755] tcp_recvmsg+0x8d2/0x9de
 [c04808dc] core_sys_select+0x218/0x2f3
 [c05dec95] inet_sendmsg+0x3b/0x45
 [c0598731] sock_aio_write+0xf6/0x102
 [c045da36] get_page_from_freelist+0x23b/0x2a5
 [c04754ee] do_sync_write+0xc7/0x10a
 [c0436e71] autoremove_wake_function+0x0/0x35
 [c0475d47] vfs_write+0xbc/0x154
 [c0476342] sys_write+0x41/0x67
 [c0404f70] syscall_call+0x7/0xb
 ===
NOTICE-xpd_bri: XBUS-01/XPD-08: D-Chan RX Bad checksum: [FA:FA=FC] (252)
NOTICE-xpd_bri: XBUS-01/XPD-08: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [78:3A=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [55:55=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [55:55=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [55:55=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [AA:AA=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [35:22=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX short frame (idx=3)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [35:22=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [22:22=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
NOTICE-xpd_bri: XBUS-00/XPD-10: D-Chan RX Bad checksum: [55:02=FC] (252)
NOTICE-xpd_bri: XBUS-00/XPD-10: Multibyte Drop: errno=-71
zaptel Disabled echo canceller because of tone (rx) on channel 4
NOTICE-xpd_bri: XBUS-00/XPD-00: D-Chan RX Bad checksum: [94:2D=FD] (253)
NOTICE-xpd_bri: XBUS-00/XPD-00: Multibyte Drop: errno=-71

 
Once the system becomes unstable the only way to get it up again is to
shutdown (not restart) pull the power and USB on the Xorcom Bri 4
devices. Then plug them back in and start the system up.
If the power and USB is not disconnected a the devices may look like
they are working fine but zaptel will not start stating that it can not
find span ** what ever it happens to fail on.
 
 
We are running the following versions
Asterisk - Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g
Zaptel - zaptel-1.2.18
XPP - version:trunk-r3965
 srcversion: 723B8A27A7E9750BB039D00

If you need any more information please let me know.
 
 
 

Nathan Dennis 
__ 
Integrated Solutions (QLD)P/LPhone: +61 (7) 4044 0300 
Direct: +61 (7) 4044
0302
124 Spence Street   Fax:+61 (7) 4041 6600
CAIRNS QLD 4868Mobile: 0418 608609

Australia 

E-mail: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] 
Web Site: www.i-solutions.net.au http://www.i-solutions.net.au/ 

Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
Sydney
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which 
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Any review, retransmission, dissemination or other use of, or taking of
any 
action in reliance upon, this information by persons or entities other
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[asterisk-users] Zap dialling issues

2007-06-27 Thread Nathan Dennis 
I'm having problems getting an Xorcom USB Bri 4 dialling out in
Australia.
 
I can receive calls into the system without an issue, but I can not for
the life of me dial out of the system. Below are my configs, I'm hoping
its something simple that I just can't see as I've been looking at it
for to long. Can any one point me in the right direction.
 
P.S. Yes it is meant to be in TE PTP mode as the current Digium B410P
works fine in that mode
 
 
 
/etc/asterisk/extensions.conf  Extract
[internal]
include=features
include=speeddial
 

;Extention number for main Q
exten = 700,1,Goto(mainq,q,1)
   
;-
;Calling a local extensions mailbox
exten = _*7XX,1,Set(Extension=${EXTEN:1})
exten = _*7XX,n,Goto(directtovoicemail,s,1)
 
;--
 
;Static externaly accessable Conference room with recording
exten =
599,1,Set(MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/conference-50
0-${EPOCH});
exten = 599,n,MeetMe(500,cMr,4081)
exten = 599,n,Hangup
 
;Dynamic Conference rooms for internal users to transfer callers to
exten = _5XX,1,MeetMe(${EXTEN},cMd)
exten = _5XX,n,Hangup
 
exten = 6000,1,Dial(zap/0418608609)
 
 
/etc/asterisk/zapata.conf
[channels]
;echocancel = yes
;transfer = yes
callgroup=1
pickupgroup=1
 
; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE 
group=0,11
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
 
; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE 
group=0,12
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
 

; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE 
;group=0,13
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 7-8
 
 
 
; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE 
;group=0,14
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 10-11
 
/etc/zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
 
# It must be in the module loading order
 

# Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3
 
# Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6
 
# Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9
 
# Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE
span=4,4,1,ccs,ami
# termtype: te
bchan=10-11
dchan=12
 
# Global data
 
loadzone= au
defaultzone = au

 
Error recieved in console without group
 
-- Executing Dial(SIP/701-09f0fc18, zap/0418608609) in new stack
Jun 27 19:27:13 NOTICE[4011]: app_dial.c:1089 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/701-09f0fc18' status is 'CONGESTION'

 
 
Error recieved in console with g0 in the dial string
 -- Executing Dial(SIP/701-08d76e98, zap/g0/0418608609) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0418608609
-- Zap/1-1 is proceeding passing it to SIP/701-08d76e98
-- Channel 0/1, span 1 got hangup request
-- Channel 0/1, span 1 received AOC-E charging 0 units
Jun 27 19:28:35 WARNING[4046]: app_dial.c:738 wait_for_answer: Unable to
forward voice
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/701-08d76e98' status is
'CHANUNAVAIL'

 

Nathan Dennis 
__ 
Integrated Solutions (QLD)P/LPhone: +61 (7) 4044 0300 
Direct: +61 (7) 4044
0302
124 Spence Street   Fax:+61 (7) 4041 6600
CAIRNS QLD 4868Mobile: 0418 608609

Australia 

E-mail: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] 
Web Site: www.i-solutions.net.au http://www.i-solutions.net.au/ 

Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
Sydney
__ 
The information transmitted is intended only for the person or entity to
which 
it is addressed and may contain confidential and/or privileged material.

Any review, retransmission, dissemination or other use of, or taking of
any 
action in reliance upon, this information by persons or entities other
than the 
intended recipient is prohibited. If you received this in error, please
contact 
the sender and delete the material from any computer.
 
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Re: [asterisk-users] Xorcom Bri 4 Port USB

2007-06-27 Thread Nathan Dennis 
Thanks Tzafrir, that did the trick.
But please note the that the bristuff patch from xorcom has broken links in it. 
It can't download asterisk using the URL in the script. Easy enough to fix by 
pointing to a known good URL.
 



From: [EMAIL PROTECTED] on behalf of Tzafrir Cohen
Sent: Mon 25/06/2007 7:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Xorcom Bri 4 Port USB



Hi

On Mon, Jun 25, 2007 at 06:38:37PM +1000, Nathan Dennis wrote:

 Hi,
I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled 
 asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.

 So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.

 The problem I'm having is that for one I get no LEDs showing if the unit
 is in TE and NT mode (not a issue for me but may have some impact on
 things) I have no errors in any logs I can see but once zaptel and
 asterisk are started up I get a lots of warnings in asterisk such as
 the following

What is the output of:

modinfo xpp | grep version

if this is something of the sort of 'r3495' then you indeed have an
older version of the driver where BRI support has not been matuire
enough and specifically leds display was not as it is today. In current
version (e.g: the one in zaptel 1.2.18/1.4.3) you will always see an
orange LED for NT or green led for TE on the port.

Please get the version of bristuff from:

http://updates.xorcom.com/astribank/bristuff/
http://updates.xorcom.com/astribank/bristuff/bristuff-0.3.0-PRE-1y-g-xr1.tar.gz

At least until we see a new version of bristuff.

and also see:

http://updates.xorcom.com/astribank/bristuff/INSTALL.html

Also, for the sake of those who will see the messages in a search:


 Jun 25 18:18:07 WARNING[2833]: chan_zap.c:2662 pri_find_dchan: No D-channels 
 available!  Using Primary channel 3 as D-channel anyway!
   == Primary D-Channel on span 2 down
 Jun 25 18:18:07 WARNING[2834]: chan_zap.c:2662 pri_find_dchan: No D-channels 
 available!  Using Primary channel 6 as D-channel anyway!
   == Primary D-Channel on span 3 down
 Jun 25 18:18:07 WARNING[2835]: chan_zap.c:2662 pri_find_dchan: No D-channels 
 available!  Using Primary channel 9 as D-channel anyway!
   == Primary D-Channel on span 1 down


This message comes from chan_zap when a span is down. If a span has
pri_{cpe,net} signalling or bri_{cpe,net} signalling (bristuff BRI ptp)
then you'll get those messages for spans that are down. If the
signalling is bri_{cpe,net}_ptmp they'll be debug messages.



 It errors for all for ports and makes no difference if I have the
 ISDN cables connected or not. I want to run in ptp mode and
 currently use a digium B410P card on the connections that work fine
 so I know that the lines work and ptp is the correct mode.


 Following are my configs. Any pointers you can give would be greatly 
 appreciated.

 We are running Fedora 7.
 Kernel Linux 2.6.21-1.3194.fc7 #1 SMP i686 i686 i386 GNU/Linux (Standard 
 Kernel with install)
 Device has jumpers all set to TE mode.


 /etc/init.d/zaptel.conf
 # Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
 span=1,1,1,ccs,ami
 # termtype: te
 bchan=1-2
 dchan=3

 # Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
 span=2,2,1,ccs,ami
 # termtype: te
 bchan=4-5
 dchan=6

 # Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
 span=3,3,1,ccs,ami
 # termtype: te
 bchan=7-8
 dchan=9

 # Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
 #span=4,4,1,ccs,ami
 # termtype: te
 #bchan=10-11
 #dchan=12

 # Global data

 loadzone= au
 defaultzone = au


 /etc/asterisk/zapata.conf
 [channels]
 ;   echocancel = yes
 ;   transfer = yes
 ;   threewaycalling = yes

 #include zapata-channels.conf


 /etc/asterisk/zapata-channels.conf

 ; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
 callerid=asreceived
 group=0
 context=from-pstn
 switchtype = euroisdn
 signalling = bri_cpe
 channel = 1-2
 callerid=
 group=
 context=default

 ; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
 callerid=asreceived
 group=0
 context=from-pstn
 switchtype = euroisdn
 signalling = bri_cpe
 channel = 4-5
 callerid=
 group=
 context=default

 ; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
 callerid=asreceived
 group=0
 context=from-pstn
 switchtype = euroisdn
 signalling = bri_cpe
 channel = 7-8
 callerid=
 group=
 context=default

 ; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
 ;callerid=asreceived
 ;group=0
 ;context=from-pstn
 ;switchtype = euroisdn
 ;signalling = bri_cpe
 ;channel = 10-11
 ;callerid=
 ;group=
 ;context=default

--
   Tzafrir Cohen  
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]  
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Xorcom Bri 4 Port USB

2007-06-25 Thread Nathan Dennis 

Hi,
   I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled 
asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.

So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.

The problem I'm having is that for one I get no LEDs showing if the unit is in 
TE and NT mode (not a issue for me but may have some impact on things) I have 
no errors in any logs I can see but once zaptel and asterisk are started up I 
get a lots of warnings in asterisk such as the following

Jun 25 18:18:07 WARNING[2833]: chan_zap.c:2662 pri_find_dchan: No D-channels 
available!  Using Primary channel 3 as D-channel anyway!
  == Primary D-Channel on span 2 down
Jun 25 18:18:07 WARNING[2834]: chan_zap.c:2662 pri_find_dchan: No D-channels 
available!  Using Primary channel 6 as D-channel anyway!
  == Primary D-Channel on span 3 down
Jun 25 18:18:07 WARNING[2835]: chan_zap.c:2662 pri_find_dchan: No D-channels 
available!  Using Primary channel 9 as D-channel anyway!
  == Primary D-Channel on span 1 down

It errors for all for ports and makes no difference if I have the ISDN cables 
connected or not. I want to run in ptp mode and currently use a digium B410P 
card on the connections that work fine so I know that the lines work and ptp is 
the correct mode.

Following are my configs. Any pointers you can give would be greatly 
appreciated.

We are running Fedora 7.
Kernel Linux 2.6.21-1.3194.fc7 #1 SMP i686 i686 i386 GNU/Linux (Standard 
Kernel with install)
Device has jumpers all set to TE mode.


/etc/init.d/zaptel.conf
# Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3

# Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6

# Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9

# Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
#span=4,4,1,ccs,ami
# termtype: te
#bchan=10-11
#dchan=12

# Global data

loadzone= au
defaultzone = au


/etc/asterisk/zapata.conf
[channels]
;   echocancel = yes
;   transfer = yes
;   threewaycalling = yes

#include zapata-channels.conf


/etc/asterisk/zapata-channels.conf

; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
callerid=
group=
context=default

; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
callerid=
group=
context=default

; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 7-8
callerid=
group=
context=default

; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
;callerid=asreceived
;group=0
;context=from-pstn
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 10-11
;callerid=
;group=
;context=default



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