RE: [asterisk-users] Test E1 channel

[James Hawks]

When you dial directly you are bypassing
the zap and just dialing an internal extension. So that is probably why dialing
directly works. As far as the cross over cable between ports 1 and 2 I have
never attempted something like that before.





James Hawks





-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Ralph Liebessohn
Sent: Friday, July
 07, 2006 6:38 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Test E1
channel



Hi guys,

I need to make a configuration to test a E1 channel, so, in the same context I
created two extensions:
exten = 555666,1,Dial(Zap/1/5556662)
exten = 5556662,1,Dial(SIP/test)

On the E1 card I linked with a cross cable the ports 1 and 2. The leds are
signaling that the connection is ok.
But when I call 555666 the calling don't goes to client SIP/test .
If I call directly 5556662 rings on SIP/test. 
Do I have to config something else to receive calls on E1 channels?
On monday the real E1 channel will be installed and I must test it.


-- 
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn 






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RE: [asterisk-users] Asterisk stops accepting calls

[James Hawks]

This happens when your * box gets out of sync with your telco provider.
The only way to currently fix it is to restart *.

Do you log your CDR's to a database? If so asterisk will wait until the call
is logged before hanging up the channel which might be too long for your
telco provider.


James Hawks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Laurent CARON
Sent: Friday, July 07, 2006 8:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk stops accepting calls

Hi,

I've got a serious problems.

I have an * box set up at a custommer office.

* seems to work well until this message appears when i try to call from 
the outside to any number managed by *

Jul  7 17:12:08 WARNING[8792]: chan_zap.c:9256 pri_dchannel: Ring 
requested on channel 0/1 already in use on span 2. Hanging up owner.

Do anyone have a clue about it, as restarting * more often than not is 
not a really good solution?

Thanks

Laurent
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[asterisk-users] Dropped Calls Need Help

[James Hawks]

We are receiving a large amount of dropped calls on our asterisk system.
After debugging I find the following line at the same time the call is
dropped. 

(DEBUG[8882] channel.c: Got a FRAME_CONTROL (5) frame on channel)

I was unable to find very much information on this message. Just a quick
background of our system:
Tyan 2882 motherboard
Dual Core Operon Processors
2 SATA Drives with Linux Software Raid
TE411P Digium Card
TDM400P Digium Card
Intel PCI Gigabit NIC
Fedora Core 4 OS
Asterisk 1.2.9.1
Zaptel 1.2.6
Libpri 1.2.3
All phones on local LAN
No missed interrupts at all
Zttest: Best: 100.00 -- Worst: 99.987793 -- Average: 99.993896


Any help would be appreciated.

Thank You
James Hawks


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RE: [asterisk-users] Troubleshooting Random PRI disconnects

[James Hawks]

Check your setting in your zapata.conf file and try the following.

busydetect=no
priindication=outofband

James Hawks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shawn Kelley
Sent: Wednesday, July 05, 2006 8:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Troubleshooting Random PRI disconnects

Hi all,
We are having an issue with Random Disconnects wit our PRI connection.
We are going into a T100P card from a Cisco IAD. 
Below is a copy of our PRI Debug. 
We will be talking on the phone and all the sudden the line will go dead. It
acts as if the remote party hungup. However, the remote party is not hanging
up.
Can anyone give me some hints on where to look for the cause of this?



    -- Zap/25-1 is ringing
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 44/0x2C) (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) ]
    -- Zap/25-1 is ringing
    -- Zap/25-1 is ringing
    -- Zap/25-1 answered Zap/1-1
 Protocol Discriminator: Q.931 (8)  len=14
 Call Ref: len= 2 (reference 44/0x2C) (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 44/0x2C) (Originator)
 Message type: CONNECT ACKNOWLEDGE (15)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 44/0x2C) (Originator)
 Message type: DISCONNECT (69)
 [08 02 82 90]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location:
Public network serving the local user (2)
  Ext: 1  Cause: Unknown (16), class = Normal Event (1) ]
-- Processing IE 8 (cs0, Cause)
    -- Channel 0/1, span 1 got hangup request
    -- Hungup 'Zap/25-1'
  == Spawn extension (from-mcleod-pri, 1693, 4) exited non-zero on 'Zap/1-1'
  == End MixMonitor Recording Zap/1-1
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 44/0x2C) (Terminator)
 Message type: RELEASE (77)
 [08 02 81 90]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location:
Private network serving the local user (1)
  Ext: 1  Cause: Unknown (16), class = Normal Event (1) ]
    -- Hungup 'Zap/1-1'
 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 44/0x2C) (Originator)
 Message type: RELEASE COMPLETE (90)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null


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[Asterisk-Users] Switchtype

[James Hawks]

Our PRI vendor is using a Nortel DMS500 switch. Which switch
type should I use. I have been using national but we are having issues with our
connectivity.



national

dms100

4ess

5ess

euroisdn

ni1

qsig





Thank You

James Hawks








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RE: [Asterisk-Users] need help troubleshooting clipping and garbledVOIP calls

[James Hawks]

Sounds like a QoS issue with your DSL provider. If you go to
http://www.bandwidth.com/tools/voipTest it might give you some insight.

James Hawks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw
Sent: Thursday, June 29, 2006 1:27 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] need help troubleshooting clipping and garbledVOIP
calls

Hello all,
I have a problem with call quality with my Asterisk setup. I'm doing VOIP 
only so far, but have a zaptel TDM400P in the box not being used. The 
problem i'm having is that when calls are placed, connected, and the far-end

is reporting that they are experiencing clipping, choppy, and garbled voice 
conversations. So bad that we have to resort to using our cell phones. This 
entire setup is still being built, but any phone attached is experiencing 
this. Call volume is almost nil (under 20 total incoming calls a day). This 
is a small business setup. The server is used exclusively for Asterisk, so 
it isn't a fileserver, or anything else.

The setup is as such:

ipphone  ---cisco 2900XL switch  Cisco 2621 router --- dsl modem 
--DSL --- VOIPprovider

I've configured the switch and the router to set priority and qos to 
prioritize voice traffic above data.
Funny thing is, there is not data REALLY hitting the network. I have setup 2

vlans, data vlan, and voice vlan. There are two work stations on the 
network, and neither is being used to hit the internet heavily (office is 
still being setup).

Any pointers or suggestions anyone have for me as to were to look for this 
poor quality?
It seems only the Far-end (called party), is hearing this and not the 
calling party.

I haven't tried switching out the phones because we only have 1 type, and 
any of the phones i used exhibit these problems. I will try softphones to 
see if it is truly a networking issue or Phone issue.

Is anyone using a cisco 2900 switch or router and care to provide config 
samples of their COS/QOS setup?

Thanks!

Terrelle Shaw


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[Asterisk-Users] txfax problem

[James Hawks]

I can not get txfax to work with a multiple page 
tiff image. It works fine with a single page tiff image. Has anyone had this 
problem before and if so is there a fix?


Thank You,James HawksCTO 
CustomerFunding.com Inc.(877) 784-1030 
x206
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Re: [Asterisk-Users] txfax problem

[James Hawks]

I am using Asterfax but I also tried the following 
with the same results:

[macro-testfax]exten = 
s,1,Set(FAXFILE=/var/spool/asterisk/fax/QWWWxx1.tif))exten = 
s,2,txfax(${FAXFILE}|caller)

[default]exten = 7001,1,Answerexten 
= 7001,n,Dial(ZAP/97/94805175098|30|gM(testfax))exten = 
7001,n,Hangupexten = 7001,102,Congestion


I did this to see if Asterfax was the problem or it 
was txfax. It seems to be txfax or the tif file. I also tried using a fax 
machine attached to my asterisk server and the fax went through with no problems 
so that eliminated a hardware issue with my Digium cards.


Thank You,James HawksCTO CustomerFunding.com Inc.(877) 784-1030 
x206

  - Original Message - 
  From: 
  Technical Support 
  
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, March 27, 2006 4:31 
PM
  Subject: RE: [Asterisk-Users] txfax 
  problem
  
  James,
  
  Which fax application are you 
  using?
  
  MD
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of James 
  HawksSent: Monday, March 27, 2006 5:29 PMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] txfax problem
  
  I can not get txfax to work with a multiple page 
  tiff image. It works fine with a single page tiff image. Has anyone had this 
  problem before and if so is there a fix? 
  
  
  Thank You,James HawksCTO 
  CustomerFunding.com Inc.(877) 784-1030 x206
  
  

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