[asterisk-users] Temporarily placing confbridge participants on hold - two way muting

[Steve Hanselman]

Is there a way of temporarily suspending participants in a conference?
 
Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the 
confbridge session that D and E can't hear, is there a way to suspend D and E 
for a while (whilst they are played music or whatever) and later join them back 
in?
 
Failing that, I was considering kicking them and using an AGI script to rejoin 
them to the conference but I wasn't sure how to do that from the script (the 
rejoin, not the kick)?
 
Any pointers or suggestions welcomed.
 
(In a nutshell it's for a situation where certain participants need to have 
privacy in the conference from a group of others and it all needs to be driven 
from an AGI script).
 
Regards
Steve
 


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Re: [asterisk-users] How to find the CDR call start time value

[Steve Hanselman]

That's exactly what I was looking for, I'd found this
http://www.voip-info.org/wiki/view/Asterisk+variables which seems to be
a partial copy of the same thing.



Thanks







From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Krunal
Patel
Sent: 02 October 2008 11:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to find the CDR call start time value



HI Steven,

You can get call start time by ${CDR(start)} .
For more information of asterisk variables , please check out
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List

Thanks,
Krunal Patel

On Thu, Oct 2, 2008 at 3:08 PM, Steve Hanselman <[EMAIL PROTECTED]>
wrote:

Can anyone suggest how I can find the value of the call start time that
will be logged by CDR in the dialplan?



I've taken a look through the variables but I can't see anything that
seems to hold this?





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[asterisk-users] How to find the CDR call start time value

[Steve Hanselman]

Can anyone suggest how I can find the value of the call start time that
will be logged by CDR in the dialplan?



I've taken a look through the variables but I can't see anything that
seems to hold this?







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Re: [asterisk-users] Lone worker system

[Steve Hanselman]

Spot on (except for the "shitty way", it's pretty standard, in building
there are paging systems that start an escalating tone, beyond the
building these don't work, so if you're away then we'd be dialling the
mobile).

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: 12 May 2008 16:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Lone worker system

He wants a randomly generated phone call to be generated to a specific
extension.
Eg once an hour for the midnight to dawn shift at a random time per
hour.

When the person picks up they are asked a question using an audio file.
(or text to speech).

Then the person has to enter the correct dtmf answering the question (eg
1 - 5)

If the person fails to answer the phone (I'm guessing here but a second
call will be placed 2 mins later).

If this call is also 'fail to answer' an escalation call to a supervisor
or something similar will occur indicating that the 'lone worker' failed
to respond and is either - dead from a stabbing, or 2 jerking off in the
bathroom and not at his post.





Cheers,
Dean 

P.S. Hope your lone worker is paid a lot to be working for a shitty
company checking up on them like that :)



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Monday, May 12, 2008 10:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Lone worker system
> 
> On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman
<[EMAIL PROTECTED]>
> wrote:
> >
> > Has anybody got any scripts for a lone worker system using Asterisk
before I
> > write them?
> >
> > Something along the lines of a regular phonecall with some kind of
random
> > question (e.g. press 1 then 5) to provide monitoring of lone workers
with
> > alerts?
> >
> > Steve
> >
> 
> I think a little more elaboration would get you more helpful advice.
> I have read your message a couple of times and still don't really
> understand what you need.
> 
> Thanks,
> Steve Totaro
> 
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[asterisk-users] Lone worker system

[Steve Hanselman]

Has anybody got any scripts for a lone worker system using Asterisk
before I write them?



Something along the lines of a regular phonecall with some kind of
random question (e.g. press 1 then 5) to provide monitoring of lone
workers with alerts?



Steve







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Re: [asterisk-users] Bridged PRI calls - processor involvement?

[Steve Hanselman]

Tracked this down (or more to the point found the issue causing it), it
was high levels of bursty disk activity.

The iowait went through the roof (30-40%).

The disks are scsi serviced by an MPT-Fusion controller in a Dell
Poweredge 2850.

We're using LVM to bind the disks into a JBOD set.

Steve

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Hanselman
Sent: 11 June 2007 10:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement?

I checked for BIOS upgrades the other week and there were none.

I'm starting to suspect kernel changes as being the reason for this so I
guess I'm going to have to remove some of the patchy disk activity to
smooth the load and then start researching!!!

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: 11 June 2007 09:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement?

On Mon, 11 Jun 2007, Steve Hanselman wrote:

> This is the io wait figure from vmstat.
>
> If I run a vmstat 2 whilst I'm on a call I can see that the "wa"
figure
> gets very high when the missing audio problem occurs.

I once looked after a Dell 2850 that exhibited some odd behaviour that I

never got to the bottom of. It would seem to lock-up or just crawl for
2-3 
seconds every now & then. Nothing logged, noting on the console. It had
6 
SCSI drives fitted. I rebuilt the server twice, rebuilt the s/w RAID 
arrays twice, even put all 6 drives in another box (which appeared
towork 
OK), but never got to the bottom of it. Each disk would benchmark really

fast individually, Ethernet performance was good, but overall, when 
everything was used together, it just didn't feel right. (compared to 
other Dells and other servers, biger & smaller that I've built and used 
over the years). I'd see processes hung in a "D" state (waiting for IO
to 
complete) for what seemed like an overly long time, (waiting on disk),
but 
...

I suspected a BIOS pproblem, but never had a chance to get to the bottom

of it. (It was a live server doing *everything* for a small company -
DNS, 
NIS, NFS, Intranet/WiKi, Samba, etc, etc, etc,... so taking it offline
for 
tests was problematic)

So I wonder if looking at the BIOS and seeing if there are any Dell 
upgrades avalable for it might help?

Gordon


  >
> Steve
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
> Fredrickson
> Sent: 08 June 2007 19:38
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Bridged PRI calls - processor
involvement?
>
> iowait time?  I'm not familiar with that.  Where are you seeing that?
> Also, is it a reproducible problem?
>
> ---
> Matthew Fredrickson
> Software Engineer
> Digium, Inc.
>
> On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote:
>
>> It probably did but we run in updates every week and nobody can state
>> exactly when the problem started only "a few weeks ago" - not very
>> helpful.
>>
>> I can see that when I hear the issue the iowait time is high on the
>> processor.
>>
>> Steve
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
>> Fredrickson
>> Sent: 08 June 2007 15:43
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Bridged PRI calls - processor
>> involvement?
>>
>> Did it accompany an update you made?  If you can find out what
version
>> the problem started occurring, that would help in fixing the problem,
>>
>> Matthew Fredrickson
>> Software/Hardware Engineer
>> Digium, Inc.
>>
>> On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote:
>>
>>> The setup.
>>>
>>> Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
>>> updates applied), the TE410 lives on it's own interrupt.
>>> Asterisk sits between our telco and a PRI enabled PBX.
>>> These are the relevant versions installed:
>>>
>>> Linux: 2.6.20-1.2316.fc5smp
>>> Zaptel: 1:1.4.2.1-34.fc5
>>> Asterisk: 1:1.4.0-34.fc5.at
>>> Libpri: 1:1.4.0-16.fc5.at
>>> Wildcard details:
>>> Found TE4XXP at base address fe3ffc00, remapped to f88bec00
>>> TE4XXP version c01a016a, burst OFF, slip debug: OFF
>>> Octasic optimized!
>>> FALC version: 0005, Board ID: 00
>>> Reg 0: 0x377bb400
>>> Reg 1: 0x377bb000
>>> Reg 2: 0xfff

RE: [asterisk-users] Bridged PRI calls - processor involvement?

[Steve Hanselman]

I checked for BIOS upgrades the other week and there were none.

I'm starting to suspect kernel changes as being the reason for this so I
guess I'm going to have to remove some of the patchy disk activity to
smooth the load and then start researching!!!

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: 11 June 2007 09:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement?

On Mon, 11 Jun 2007, Steve Hanselman wrote:

> This is the io wait figure from vmstat.
>
> If I run a vmstat 2 whilst I'm on a call I can see that the "wa"
figure
> gets very high when the missing audio problem occurs.

I once looked after a Dell 2850 that exhibited some odd behaviour that I

never got to the bottom of. It would seem to lock-up or just crawl for
2-3
seconds every now & then. Nothing logged, noting on the console. It had
6
SCSI drives fitted. I rebuilt the server twice, rebuilt the s/w RAID
arrays twice, even put all 6 drives in another box (which appeared
towork
OK), but never got to the bottom of it. Each disk would benchmark really

fast individually, Ethernet performance was good, but overall, when
everything was used together, it just didn't feel right. (compared to
other Dells and other servers, biger & smaller that I've built and used
over the years). I'd see processes hung in a "D" state (waiting for IO
to
complete) for what seemed like an overly long time, (waiting on disk),
but
...

I suspected a BIOS pproblem, but never had a chance to get to the bottom

of it. (It was a live server doing *everything* for a small company -
DNS,
NIS, NFS, Intranet/WiKi, Samba, etc, etc, etc,... so taking it offline
for
tests was problematic)

So I wonder if looking at the BIOS and seeing if there are any Dell
upgrades avalable for it might help?

Gordon


  >
> Steve
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
> Fredrickson
> Sent: 08 June 2007 19:38
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Bridged PRI calls - processor
involvement?
>
> iowait time?  I'm not familiar with that.  Where are you seeing that?
> Also, is it a reproducible problem?
>
> ---
> Matthew Fredrickson
> Software Engineer
> Digium, Inc.
>
> On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote:
>
>> It probably did but we run in updates every week and nobody can state
>> exactly when the problem started only "a few weeks ago" - not very
>> helpful.
>>
>> I can see that when I hear the issue the iowait time is high on the
>> processor.
>>
>> Steve
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
>> Fredrickson
>> Sent: 08 June 2007 15:43
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Bridged PRI calls - processor
>> involvement?
>>
>> Did it accompany an update you made?  If you can find out what
version
>> the problem started occurring, that would help in fixing the problem,
>>
>> Matthew Fredrickson
>> Software/Hardware Engineer
>> Digium, Inc.
>>
>> On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote:
>>
>>> The setup.
>>>
>>> Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
>>> updates applied), the TE410 lives on it's own interrupt.
>>> Asterisk sits between our telco and a PRI enabled PBX.
>>> These are the relevant versions installed:
>>>
>>> Linux: 2.6.20-1.2316.fc5smp
>>> Zaptel: 1:1.4.2.1-34.fc5
>>> Asterisk: 1:1.4.0-34.fc5.at
>>> Libpri: 1:1.4.0-16.fc5.at
>>> Wildcard details:
>>> Found TE4XXP at base address fe3ffc00, remapped to f88bec00
>>> TE4XXP version c01a016a, burst OFF, slip debug: OFF
>>> Octasic optimized!
>>> FALC version: 0005, Board ID: 00
>>> Reg 0: 0x377bb400
>>> Reg 1: 0x377bb000
>>> Reg 2: 0x
>>> Reg 3: 0x
>>> Reg 4: 0x0001
>>> Reg 5: 0x
>>> Reg 6: 0xc01a016a
>>> Reg 7: 0x1f00
>>> Reg 8: 0x
>>> Reg 9: 0x00ff
>>> Reg 10: 0x004a
>>> TTE4XXP: Launching card: 0
>>> TE4XXP: Setting up global serial parameters
>>> Found a Wildcard: Wildcard TE410P (3rd Gen)
>>> TE4XXP: Span 1 configured for CCS/HDB3/CRC4
>>> TE4XXP: Span 2 configured for CCS/HDB3/CRC4
>>>
>>>
>>>
>>> The pr

RE: [asterisk-users] Bridged PRI calls - processor involvement?

[Steve Hanselman]

Hi Steve,

No, nothing like that, it has various updated from an 8mb Internet link
and that's about it, I feel now that it's more down to disk I/O with the
mpt driver than network.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: 09 June 2007 13:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement?

Are you running recording on your box or FTPing large recording files or
PDFs or anything other than just voice traffic?  Has voice traffic
spiked in conjunction with your problems?

Are you doing any kind of port monitoring/mirroring on your switch?
Most people look at the 100mb or 1Gb figure but there is also another
very important spec to look at when evaluating a switch.  It is Frame
Forwarding Rate measured by Mpps.  Take a look at your switch's docs and
let us know what your FFR is and if you are doing any mirroring or link
aggregation.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Hanselman
> Sent: Friday, June 08, 2007 12:23 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Bridged PRI calls - processor
involvement?
>
> It probably did but we run in updates every week and nobody can state
> exactly when the problem started only "a few weeks ago" - not very
> helpful.
>
> I can see that when I hear the issue the iowait time is high on the
> processor.
>
> Steve
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
> Fredrickson
> Sent: 08 June 2007 15:43
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Bridged PRI calls - processor
involvement?
>
> Did it accompany an update you made?  If you can find out what version
> the problem started occurring, that would help in fixing the problem.
>
> Matthew Fredrickson
> Software/Hardware Engineer
> Digium, Inc.
>
> On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote:
>
> > The setup.
> >
> > Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
> > updates applied), the TE410 lives on it's own interrupt.
> > Asterisk sits between our telco and a PRI enabled PBX.
> > These are the relevant versions installed:
> >
> > Linux: 2.6.20-1.2316.fc5smp
> > Zaptel: 1:1.4.2.1-34.fc5
> > Asterisk: 1:1.4.0-34.fc5.at
> > Libpri: 1:1.4.0-16.fc5.at
> > Wildcard details:
> > Found TE4XXP at base address fe3ffc00, remapped to f88bec00
> > TE4XXP version c01a016a, burst OFF, slip debug: OFF
> > Octasic optimized!
> > FALC version: 0005, Board ID: 00
> > Reg 0: 0x377bb400
> > Reg 1: 0x377bb000
> > Reg 2: 0x
> > Reg 3: 0x
> > Reg 4: 0x0001
> > Reg 5: 0x
> > Reg 6: 0xc01a016a
> > Reg 7: 0x1f00
> > Reg 8: 0x
> > Reg 9: 0x00ff
> > Reg 10: 0x004a
> > TTE4XXP: Launching card: 0
> > TE4XXP: Setting up global serial parameters
> > Found a Wildcard: Wildcard TE410P (3rd Gen)
> > TE4XXP: Span 1 configured for CCS/HDB3/CRC4
> > TE4XXP: Span 2 configured for CCS/HDB3/CRC4
> >
> >
> >
> > The problem:
> >
> > At random points during calls we lose 1-3 seconds of speech (both
ways
> > both callee and caller), this can be replicated (or at least a very
> > good
> > approximation!) by generating a high level of interrupt/cpu activity
> > (for instance copying data from a USB caddy as we tried the other
day
> > in
> > an attempt to reproduce this more reliably).
> >
> > The calls are bridged PRI:PRI calls, no VOIP involvement.
> >
> > This was not a problem until approx 3-4 weeks ago, but I can't tie
it
> > down to an exact date.
> >
> > Steve
> >
> >
> >> Interrupt sharing is not a problem anymore with those cards.  What
> >> version of zaptel did you try installing?  Can you explain more
about
> >> your problems?  Also, your configuration and setup would help out
as
> >> well.
> >>
> >> ---
> >> Matthew Fredrickson
> >> Digium, Inc.
> >
> >
> > The information contained in this email is intended for the personal
> > and confidential use
> > of the addressee only. It may also be privileged information. If you
> > are not the intended
> > recipient then you are hereby notified that you have received this
> > document in error and
> > that any re

RE: [asterisk-users] Bridged PRI calls - processor involvement?

[Steve Hanselman]

This is the io wait figure from vmstat.

If I run a vmstat 2 whilst I'm on a call I can see that the "wa" figure
gets very high when the missing audio problem occurs.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: 08 June 2007 19:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement?

iowait time?  I'm not familiar with that.  Where are you seeing that?  
Also, is it a reproducible problem?

---
Matthew Fredrickson
Software Engineer
Digium, Inc.

On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote:

> It probably did but we run in updates every week and nobody can state
> exactly when the problem started only "a few weeks ago" - not very
> helpful.
>
> I can see that when I hear the issue the iowait time is high on the
> processor.
>
> Steve
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
> Fredrickson
> Sent: 08 June 2007 15:43
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Bridged PRI calls - processor
> involvement?
>
> Did it accompany an update you made?  If you can find out what version
> the problem started occurring, that would help in fixing the problem.
>
> Matthew Fredrickson
> Software/Hardware Engineer
> Digium, Inc.
>
> On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote:
>
>> The setup.
>>
>> Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
>> updates applied), the TE410 lives on it's own interrupt.
>> Asterisk sits between our telco and a PRI enabled PBX.
>> These are the relevant versions installed:
>>
>> Linux: 2.6.20-1.2316.fc5smp
>> Zaptel: 1:1.4.2.1-34.fc5
>> Asterisk: 1:1.4.0-34.fc5.at
>> Libpri: 1:1.4.0-16.fc5.at
>> Wildcard details:
>> Found TE4XXP at base address fe3ffc00, remapped to f88bec00
>> TE4XXP version c01a016a, burst OFF, slip debug: OFF
>> Octasic optimized!
>> FALC version: 0005, Board ID: 00
>> Reg 0: 0x377bb400
>> Reg 1: 0x377bb000
>> Reg 2: 0x
>> Reg 3: 0x
>> Reg 4: 0x0001
>> Reg 5: 0x
>> Reg 6: 0xc01a016a
>> Reg 7: 0x1f00
>> Reg 8: 0x
>> Reg 9: 0x00ff
>> Reg 10: 0x004a
>> TTE4XXP: Launching card: 0
>> TE4XXP: Setting up global serial parameters
>> Found a Wildcard: Wildcard TE410P (3rd Gen)
>> TE4XXP: Span 1 configured for CCS/HDB3/CRC4
>> TE4XXP: Span 2 configured for CCS/HDB3/CRC4
>>
>>
>>
>> The problem:
>>
>> At random points during calls we lose 1-3 seconds of speech (both
ways
>> both callee and caller), this can be replicated (or at least a very
>> good
>> approximation!) by generating a high level of interrupt/cpu activity
>> (for instance copying data from a USB caddy as we tried the other day
>> in
>> an attempt to reproduce this more reliably).
>>
>> The calls are bridged PRI:PRI calls, no VOIP involvement.
>>
>> This was not a problem until approx 3-4 weeks ago, but I can't tie it
>> down to an exact date.
>>
>> Steve
>>
>>
>>> Interrupt sharing is not a problem anymore with those cards.  What
>>> version of zaptel did you try installing?  Can you explain more
about
>>> your problems?  Also, your configuration and setup would help out as
>>> well.
>>>
>>> ---
>>> Matthew Fredrickson
>>> Digium, Inc.
>>
>>
>> The information contained in this email is intended for the personal
>> and confidential use
>> of the addressee only. It may also be privileged information. If you
>> are not the intended
>> recipient then you are hereby notified that you have received this
>> document in error and
>> that any review, distribution or copying of this document is strictly
>> prohibited. If you have
>> received  this communication in error, please notify Brendata
>> immediately on:
>>
>> +44 (0)1268 466100, or email '[EMAIL PROTECTED]'
>>
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>> Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UK
>> Registered Office as above. Registered in England No. 2764339
>>
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RE: [asterisk-users] Bridged PRI calls - processor involvement?

[Steve Hanselman]

It probably did but we run in updates every week and nobody can state
exactly when the problem started only "a few weeks ago" - not very
helpful.

I can see that when I hear the issue the iowait time is high on the
processor.

Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: 08 June 2007 15:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement?

Did it accompany an update you made?  If you can find out what version 
the problem started occurring, that would help in fixing the problem.

Matthew Fredrickson
Software/Hardware Engineer
Digium, Inc.

On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote:

> The setup.
>
> Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
> updates applied), the TE410 lives on it's own interrupt.
> Asterisk sits between our telco and a PRI enabled PBX.
> These are the relevant versions installed:
>
> Linux: 2.6.20-1.2316.fc5smp
> Zaptel: 1:1.4.2.1-34.fc5
> Asterisk: 1:1.4.0-34.fc5.at
> Libpri: 1:1.4.0-16.fc5.at
> Wildcard details:
> Found TE4XXP at base address fe3ffc00, remapped to f88bec00
> TE4XXP version c01a016a, burst OFF, slip debug: OFF
> Octasic optimized!
> FALC version: 0005, Board ID: 00
> Reg 0: 0x377bb400
> Reg 1: 0x377bb000
> Reg 2: 0x
> Reg 3: 0x
> Reg 4: 0x0001
> Reg 5: 0x
> Reg 6: 0xc01a016a
> Reg 7: 0x1f00
> Reg 8: 0x
> Reg 9: 0x00ff
> Reg 10: 0x004a
> TTE4XXP: Launching card: 0
> TE4XXP: Setting up global serial parameters
> Found a Wildcard: Wildcard TE410P (3rd Gen)
> TE4XXP: Span 1 configured for CCS/HDB3/CRC4
> TE4XXP: Span 2 configured for CCS/HDB3/CRC4
>
>
>
> The problem:
>
> At random points during calls we lose 1-3 seconds of speech (both ways
> both callee and caller), this can be replicated (or at least a very
> good
> approximation!) by generating a high level of interrupt/cpu activity
> (for instance copying data from a USB caddy as we tried the other day
> in
> an attempt to reproduce this more reliably).
>
> The calls are bridged PRI:PRI calls, no VOIP involvement.
>
> This was not a problem until approx 3-4 weeks ago, but I can't tie it
> down to an exact date.
>
> Steve
>
>
>> Interrupt sharing is not a problem anymore with those cards.  What
>> version of zaptel did you try installing?  Can you explain more about
>> your problems?  Also, your configuration and setup would help out as
>> well.
>>
>> ---
>> Matthew Fredrickson
>> Digium, Inc.
>
>
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RE: [asterisk-users] Bridged PRI calls - processor involvement?

[Steve Hanselman]

The setup.

Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
updates applied), the TE410 lives on it's own interrupt.
Asterisk sits between our telco and a PRI enabled PBX.
These are the relevant versions installed:

Linux: 2.6.20-1.2316.fc5smp
Zaptel: 1:1.4.2.1-34.fc5
Asterisk: 1:1.4.0-34.fc5.at
Libpri: 1:1.4.0-16.fc5.at
Wildcard details:
Found TE4XXP at base address fe3ffc00, remapped to f88bec00
TE4XXP version c01a016a, burst OFF, slip debug: OFF
Octasic optimized!
FALC version: 0005, Board ID: 00
Reg 0: 0x377bb400
Reg 1: 0x377bb000
Reg 2: 0x
Reg 3: 0x
Reg 4: 0x0001
Reg 5: 0x
Reg 6: 0xc01a016a
Reg 7: 0x1f00
Reg 8: 0x
Reg 9: 0x00ff
Reg 10: 0x004a
TTE4XXP: Launching card: 0
TE4XXP: Setting up global serial parameters
Found a Wildcard: Wildcard TE410P (3rd Gen)
TE4XXP: Span 1 configured for CCS/HDB3/CRC4
TE4XXP: Span 2 configured for CCS/HDB3/CRC4



The problem:

At random points during calls we lose 1-3 seconds of speech (both ways
both callee and caller), this can be replicated (or at least a very good
approximation!) by generating a high level of interrupt/cpu activity
(for instance copying data from a USB caddy as we tried the other day in
an attempt to reproduce this more reliably).

The calls are bridged PRI:PRI calls, no VOIP involvement.

This was not a problem until approx 3-4 weeks ago, but I can't tie it
down to an exact date.

Steve


> Interrupt sharing is not a problem anymore with those cards.  What
> version of zaptel did you try installing?  Can you explain more about
> your problems?  Also, your configuration and setup would help out as 
> well.
>
> ---
> Matthew Fredrickson
> Digium, Inc.


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[asterisk-users] Bridged PRI calls - processor involvement?

[Steve Hanselman]

On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does 
the processor have?

We're now seeing chunks of missing audio and I can't tell whether this is due 
to a kernel upgrade or to a zaptel/libpri/asterisk upgrade.

I'm not seeing missed interrupts (from a cat of the proc/zaptel files), any 
other ideas on how I could go about tracking this down?

I'm thinking of enabling the debug options on the module to see if this can 
throw any further light on the problem.

As I say, all was fine, the te410 lives on its own interrupt and all was fine 
until a few weeks back.



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RE: [asterisk-users] Cutted audio or 2/3s blankson EuroISDN- Asterisk1.4

[Steve Hanselman]

We're running 1.4.0 of asterisk
1.4.2.1 of zaptel
And kernel 2.6.20-1.2316.fc5smp



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RE: [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4

[Steve Hanselman]

We're also seeing the same thing, our calls are bridged zaptel calls between 
ISDN30 PRI interfaces on a single TE410P.

We don't' appear to have any lost interrupts.

Same as stated, 2-3 second gaps in audio.

Steve




-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Administrator 
TOOTAI
Sent: 11 May 2007 09:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- 
Asterisk1.4

Steve Totaro a écrit :

Hi Steve
> Your Zap conf files would be helpful.  Zttest results?  Cat
> /proc/interrupts.  Sharing interrupts?
No. Zap con files should not be relevant as we are using ISDN.

[EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/zaptel.conf

loadzone = us
defaultzone=us

[EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the Zap channel
; CLI> reload chan_zap.so
;   will reload the configuration file,
;   but not all configuration options are
;   re-configured during a reload.

[trunkgroups]

[channels]
;
context=default
;
switchtype=national
;
signalling=fxo_ls
;
rxwink=300  ; Atlas seems to use long (250ms) winks
;
usecallerid=yes
;
hidecallerid=no
;
callwaiting=yes
;
usecallingpres=yes
;
callwaitingcallerid=yes
;
threewaycalling=yes
;
transfer=yes
;
canpark=yes
;
cancallforward=yes
;
callreturn=yes
;
echocancel=yes
;
echocancelwhenbridged=yes
;
rxgain=0.0
txgain=0.0
;
group=1
; make these both the same.  Groups range from 0 to 63.
;
callgroup=1
pickupgroup=1
;
immediate=no


[EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /proc/interrupts
   CPU0   CPU1   CPU2   CPU3
  0:  109917508  0  0  0IO-APIC-edge  timer
  1:  12365  0  0  0IO-APIC-edge  i8042
  8:  444560118  0  0  0IO-APIC-edge  rtc
  9:  0  0  0  0   IO-APIC-level  acpi
12:  11367  0  0  0IO-APIC-edge  i8042
14:3944731  0  0  0IO-APIC-edge  ide0
58:  0  0  0  0   IO-APIC-level
uhci_hcd:usb1, uhci_hcd:usb3, ehci_hcd:usb5
66:  0  0  0  0   IO-APIC-level
uhci_hcd:usb2, uhci_hcd:usb4
74:4552211  0  0  0   IO-APIC-level  libata
90:   18418187  0  0  0 PCI-MSI  eth0
98:   27358592  0  0  0   IO-APIC-level  HFC-multi
106:   27358571  0  0  0   IO-APIC-level  HFC-multi
NMI:  14333691827   1273
LOC:  109917988  109917975  109917950  109917910
ERR:  0
MIS:  0

We use ztdummy for Meetme:

[EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ sudo ./zttest
Opened pseudo zap interface, measuring accuracy...
99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965%
99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.951172%
99.938965% 99.963379%
99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965%
99.963379% 99.963379%
99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379%
99.963379% 99.938965%
99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379%
99.938965% 99.963379%
99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965%
99.963379% 99.938965%
99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379%
99.938965% 99.951172%
99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965%
99.963379% 99.963379%
99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379%
99.963379% 99.938965%
99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379%
99.938965% 99.963379%
99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965%
99.963379% 99.938965%
--- Results after 87 passes ---
Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952721

lsmod, zttranscode was loaded, I remove it:

[EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ lsmod
Module  Size  Used by
ztdummy10056  0
tcp_diag6400  0
inet_diag  16784  1 tcp_diag
mISDN_dsp 201384  1
hfcmulti   79884  1
mISDN_capi107116  1
l3udss146744  1
mISDN_l2   44616  1
mISDN_l1   17560  1
mISDN_core 88224  6
mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1
capi   23616  0
capifs 11152  2 capi
kernelcapi 56640  2 mISDN_capi,capi
zaptel197608  7 ztdummy
crc_ccitt   6784  1 zaptel
ipv6  285664  34
ppdev  14088  0
parport_pc 41640  0
lp 17736  0
parport44684  3 ppdev,parport_pc,lp
button 12192  0
ac 1

RE: [asterisk-users] Audio going blank for a few seconds andthencomesback. What could be the reason?

[Steve Hanselman]

There seem to be two threads here that mention multi-second loss with
the common part being a PRI, certainly for my situation it's purely PRI
as the asterisk box sits in between the telco and another PRI enabled
PBX and the calls are bridged between the two.

There is no network traffic involved in this case.

Not sure where to go with mine though, the load average is nice and low,
I don't see any missed interrupts and it's only started happening in the
last few weeks since an asterisk upgrade.

Latest FC6 kernel, latest yum'd asterisk, zaptel etc

Not sure whether it's worth pulling a SVN version down and building
that, the only issue is I can't currently reproduce this on demand.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: 01 June 2007 14:36
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Audio going blank for a few seconds
andthencomesback. What could be the reason?

On Friday 01 June 2007 9:24 am, Rob Schall wrote:
> comcast high-speed, thinking that would be more than enough. Turned
out
> though, with most high speed solutions, there is some limited packet
> loss and its just to be expected. You internet browsers, etc, would

Limited packet loss != **EIGHT SECONDS** of network breakage.  Jitter
buffers
and PLC takes care of most normal network indiscretions, but period
dropouts
of that big of a time aren't normal and indicate a bigger issue, either
with
the hardware or the link itself.

> normally just re-request the packet and move on, but with a stream,
> you're out of luck. The only real solution is to have a dedicated T1
or
> mpls connection or something like that for perfect quality. We have
> solid connections between our offices and haven't had a problem yet.

I have numerous installations using standard telco (Bell Canada and
Telus)
DSL, and at least one on Rogers cable here in Ontario.  No real
problems.
The odd problem if the pipe gets saturated but careful design and
monitoring
can take care of most of these problems.

I agree with Mr. Hanselman; get a packet logger on the link and see
what's
really going on.  Until that's done, everything here is just
speculation.  I
have seen bugs in the IAX2 and SIP jitter buffers on Asterisk which
cause
dropouts like this, and I'd like to see what's actually going on before
pointing any fingers.

-A.
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RE: [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?

[Steve Hanselman]

You can use tcpdump or ethereal (wireshark now) to capture the stream
and then see if there was loss during the call, just leave a capture
going then get your users to mark out the time at which they encountered
the silence, compare this to the server time (e.g. their watch to the
server) to get a time difference, then figure out what time you need to
look at in the trace.





Steve





From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent: 01 June 2007 13:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Audio going blank for a few seconds
andthencomes back. What could be the reason?



There are some remote extensions connected on this system, and calling
long distance is purely on voip. These remote extensions also face the
same thing, i.e. audio going blank for a few seconds, when dialing long
distance. So in this case, no PRI is involved. Its either the server, or
the network. Now I don't know how to find out what is it and why?

On 6/1/07, Steve Hanselman <[EMAIL PROTECTED]> wrote:

I think this is more related to the PRI, we've been seeing this for a
few weeks now, and our environment is bridged PRI-PRI on the same board,



Steve





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RE: [asterisk-users] Audio going blank for a few seconds and thencomes back. What could be the reason?

[Steve Hanselman]

I think this is more related to the PRI, we've been seeing this for a
few weeks now, and our environment is bridged PRI-PRI on the same board,



Steve







From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent: 10 May 2007 01:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Audio going blank for a few seconds and
thencomes back. What could be the reason?



I have Grandstream and Aastra phones. It happens on both of them.



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RE: [asterisk-users] Trouble with rxfax multi-page printing with cups

[Steve Hanselman]

Fax2ps is what we use, works fine.

Yum tells me it comes from libtiff

Steve

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: 08 September 2006 11:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Trouble with rxfax multi-page printing with
cups

Hello,

Cups unfortunately don't support multi-page tiff printing.
http://www.cups.org/str.php?L1117

I have tried tiff2ps and tiff2pdf but both just embed original tiff file
and give the first page only.

Is there any solution for printing multi-page tiff easily? More likely
an alternative lp than bash or any other script.

bye,
Zsolt
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RE: [asterisk-users] Unable to match on CallerID in an include block

[Steve Hanselman]

I'll run some more tests but it's not very different from the posting?

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 18:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to match on CallerID in an include
block

I suspect that your dialplan is more than you show ;)

It works just fine for me with svn trunk

[from-sip]

include => common

[common]
exten => 1234,1,NoOp(Hmm ${CALLERID(num)})

exten => 1234/7708,1,NoOp(Here)

If I dial 1234 from my 7708 extension, I get the NoOp(Here)
If I dial 1234 from my 7701 extension, I get the NoOp(Hmm 7701)

Julian.

Steve Hanselman wrote:
> Hi Julian,
>
> Ah, a very good point, I put that in my first cut but had completely
> forgotten in this one!
>
> 1.2.10
>
> Steve
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Julian
> Lyndon-Smith
> Sent: 22 August 2006 17:30
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Unable to match on CallerID in an
include
> block
>
> What version of asterisk ?
>
> Julian
>
> Steve Hanselman wrote:
>> Is there any reason why I can't use the xxx/callerid format in an
>> include section?
>>
>>
>>
>> It doesn't seem to work, but if I paste the lines into the main
> section
>> where I include the block it does?
>>
>>
>>
>>
>>
>>
>>
>> E.g. this doesn't work
>>
>>
>>
>> [telewest]
>>
>>
>>
>> Include => spamblock
>>
>>
>>
>> [spamblock]
>>
>>
>>
>> _X./12345,s,macro(spamcall)
>>
>>
>>
>> Whereas this does:
>>
>>
>>
>> [telewest]
>>
>>
>>
>> _X./12345,s,macro(spamcall)
>>
>>
>>
>>
>>
>> Any ideas?
>>
>>
>>
>> Steve
>>
>>
>>
>> The information contained in this email is intended for the personal
> and
>> confidential use
>> of the addressee only. It may also be privileged information. If you
> are
>> not the intended
>> recipient then you are hereby notified that you have received this
>> document in error and
>> that any review, distribution or copying of this document is strictly

>> prohibited. If you have
>> received this communication in error, please notify Brendata
> immediately
>> on:
>>
>> +44 (0)1268 466100, or email '[EMAIL PROTECTED]'
>>
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>> Registered Office as above. Registered in England No. 2764339
>>
>> See our current vacancies at www.brendataco.uk
>>
>>
>>
>

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The informat

RE: [asterisk-users] Unable to match on CallerID in an include block

[Steve Hanselman]

Hi Julian,

Ah, a very good point, I put that in my first cut but had completely
forgotten in this one!

1.2.10

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 17:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to match on CallerID in an include
block

What version of asterisk ?

Julian

Steve Hanselman wrote:
> Is there any reason why I can't use the xxx/callerid format in an
> include section?
>
>
>
> It doesn't seem to work, but if I paste the lines into the main
section
> where I include the block it does?
>
>
>
>
>
>
>
> E.g. this doesn't work
>
>
>
> [telewest]
>
>
>
> Include => spamblock
>
>
>
> [spamblock]
>
>
>
> _X./12345,s,macro(spamcall)
>
>
>
> Whereas this does:
>
>
>
> [telewest]
>
>
>
> _X./12345,s,macro(spamcall)
>
>
>
>
>
> Any ideas?
>
>
>
> Steve
>
>
>
> The information contained in this email is intended for the personal
and
> confidential use
> of the addressee only. It may also be privileged information. If you
are
> not the intended
> recipient then you are hereby notified that you have received this
> document in error and
> that any review, distribution or copying of this document is strictly
> prohibited. If you have
> received this communication in error, please notify Brendata
immediately
> on:
>
> +44 (0)1268 466100, or email '[EMAIL PROTECTED]'
>
> Brendata (UK) Ltd
> Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK
> Registered Office as above. Registered in England No. 2764339
>
> See our current vacancies at www.brendataco.uk
>
>
>

>
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[asterisk-users] Unable to match on CallerID in an include block

[Steve Hanselman]

Is there any reason why I can’t use the xxx/callerid
format in an include section?

 

It doesn’t seem to work, but if I paste the lines into
the main section where I include the block it does?

 

 

 

E.g. this doesn’t work

 

[telewest]

 

Include => spamblock

 

[spamblock]

 

_X./12345,s,macro(spamcall)

 

Whereas this does:

 

[telewest]

 

_X./12345,s,macro(spamcall)

 

 

Any ideas?

 

Steve

 






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RE: [asterisk-users] rx_fax problem

[Steve Hanselman]

Rxfax has no ECM, try hylafax and iaxmodem.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paradise
Dove
Sent: 01 August 2006 21:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] rx_fax problem

hi,
rx_fax fails to get fax on a bit noisy lines
but real fax devices can do that on the same line
with no problem!
what's the problem?

thanks
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RE: [Asterisk-Users] sangoma card test

[Steve Hanselman]

Create yourself a crossover cable and loop the spans, set one to provide
clock and you should quickly see them come up, this will provide a very
basic test of hardware.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 16 June 2006 10:20
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] sangoma card test

Is there any way of running a diagnostic on a Sangoma A102 card ? Our
lines have gone down and I want to avoid the usual BT "It must be your 
equipment" line with technical proof.

At the moment I have a BLUE/RED alarm on zap show status

Julian.
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RE: [Asterisk-Users] FAX over PRI

[Steve Hanselman]

We get the occasional bad fax, but it
really is an occasional one, other than that, it’s fine.

 

We don’t get any CRC errors or clock
slips on the PRI, I’d certainly say that it would be a good starting
point to check the counters on these, I’d also check that your drives are
using DMA depending on your hardware, we had a customer a while ago who ended
up doing a self install and none of his drives were enabled for DMA.

 

Steve

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Christensen
Sent: 19 May 2006 17:18
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FAX
over PRI



 

I have had nothing but
problems receiving faxes over PRIs with spandsp.  I currently have 4
systems, 4 PRIs from 4 different providers... none of them get better than 50%
success rates receiving faxes in spandsp, I constantly get cut off pages. 
No body seems to have a fix for it, and it is really frustrating. 
Supposedly it is caused by "frame slips" on the PRI, but if that is
the case, I am 4 for 4 getting crappy PRIs that can't keep time. 

These same boxes work fine when receiving faxes over fxo ports, or if I plug a
fax machine into an fxs port and call in to a spandsp extension the fax will be
received just fine, so I am left thinking it must be the PRIs, but if all PRIs
are this bad, how can anybody be using them? 

Tom



On 5/19/06, Steve
 Hanselman <[EMAIL PROTECTED]>
wrote:

Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.

We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax. 

So what you want to do is fine and will work.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto: [EMAIL PROTECTED]]
On Behalf Of Michael
Gaudette
Sent: 21 March 2006 20:34
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Subject: RE: [Asterisk-Users] FAX over PRI

Hmmm, Im not so sure I can apply this to me though.  I just want to
do
Fax-To-Email using PRI channels as the incoming lines.  Not so much
transfer
to a real fax. 

I am assuming that this is easily done with Asterisk? (I did it before
with
Asterisk SIP, but it only worked once every 10 tries or so)

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Andrew
Kohlsmith
Sent: March 21, 2006 3:25 PM 
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FAX over PRI

On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
> How should I consider Fax over PRI channels with Asterisk?  Is
the 
> quality and reliability good, or should I be prepared for alot of
grief?

I'm having good success doing fax over PRI using a TE405; one span to
the
PRI, the other to an FXS channel bank that is almost obscenely 
underutilized
(3 channels).

I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
link
is a 1-hop SDSL (VOIP only) data link.  This works well too.

-A.
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RE: [Asterisk-Users] FAX over PRI

[Steve Hanselman]

Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.

We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax.

So what you want to do is fine and will work.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Gaudette
Sent: 21 March 2006 20:34
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FAX over PRI

Hmmm, Im not so sure I can apply this to me though.  I just want to do
Fax-To-Email using PRI channels as the incoming lines.  Not so much
transfer
to a real fax.

I am assuming that this is easily done with Asterisk? (I did it before
with
Asterisk SIP, but it only worked once every 10 tries or so)

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: March 21, 2006 3:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FAX over PRI

On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
> How should I consider Fax over PRI channels with Asterisk?  Is the
> quality and reliability good, or should I be prepared for alot of
grief?

I'm having good success doing fax over PRI using a TE405; one span to
the
PRI, the other to an FXS channel bank that is almost obscenely
underutilized
(3 channels).

I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
link
is a 1-hop SDSL (VOIP only) data link.  This works well too.

-A.
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[Asterisk-Users] asterisk no longer compiles on gcc 2.95

[Steve Hanselman]

Throwing errors
relating to utils.h:
 
/usr/include/asterisk/strings.h:264: parse error before
`__extension__'/usr/include/asterisk/strings.h:264: parse error before
`;'/usr/include/asterisk/strings.h:264: warning: type defaults to `int' in
declaration of `__retval'/usr/include/asterisk/strings.h:264: `__len'
undeclared here (not in a function)/usr/include/asterisk/strings.h:264:
warning: initialization makes integer from pointer without a
cast/usr/include/asterisk/strings.h:264: initializer element is not
constant/usr/include/asterisk/strings.h:264: warning: data definition has no
type or storage class/usr/include/asterisk/strings.h:264: parse error before
`if'/usr/include/asterisk/strings.h:264: warning: type defaults to `int' in
declaration of `__retval'/usr/include/asterisk/strings.h:264: redefinition
of `__retval'/usr/include/asterisk/strings.h:264: `__retval' previously
defined here/usr/include/asterisk/strings.h:264: parse error before
`const'/usr/include/asterisk/strings.h:264: warning: data definition has no
type or storage class/usr/include/asterisk/strings.h:264: warning: type
defaults to `int' in declaration of
`__retval'/usr/include/asterisk/strings.h:264: warning: data definition has
no type or storage class/usr/include/asterisk/strings.h:264: parse error
before `}'/usr/include/asterisk/strings.h:276: conflicting types for
`strtoq'/usr/include/stdlib.h:328: previous declaration of `strtoq'In
file included from /usr/include/asterisk/module.h:35,
 
I seem to recall
there being a statement that gcc >= 3.00 is required, is this now being
enforced?
 
Steve
 
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RE: [Asterisk-Users] AgentCallbackLogin pre-# announcement?

[Steve Hanselman]

Yes, there is a patch for this (search mantis), it's static in that it's
a single announcement that doesn't currently relate to the queue.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 03 January 2006 10:00
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AgentCallbackLogin pre-# announcement?

Is there a way to have AgentCallbackLogin make an announcement before  
requiring the callee to press #?

I can not find anything in the documentation or other sites (voip-
info etc). And at the moment the way i have it setup
AgentCallbackLogin calls the agent and waits till # is pressed, it
then plays the queue greeting.

What i would like is for AgentCallbackLogin to play an announcement
before requiring # so the agent can decide wether to answer the call
based on time of day/workload etc.

Example: Agent gets a call back and when answered they hear "you have  
a sales/support/billing call, please press # to accept"

Is this possible?

Thanks
Adam
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RE: [Asterisk-Users] CID lookup from an Exchange Public folder

[Steve Hanselman]

My point exactly.

I'll take a look at that script though, if I could automate that each
night then it might be fine, tag the imports, clear out those then
re-import again.

Thanks

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: 17 December 2005 18:29
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CID lookup from an Exchange Public folder

Exchange contacts != AD entries. Contacts in Exchange are basically
email
messages with metadata.  Now, if all of your contacts WERE in AD, you
could
do a script to query AD through LDAP (that's what AD is - LDAP with MS
extensions) and you would solve latency problems when Asterisk would
query
AD instead of clunky MAPI. Here's a cool script to export contacts in a
public folder to AD:

http://www.msexchange.org/articles/Migrating-Contacts-Distribution-Lists
-Out
look-Active-Directory.html

The problem with this is maintenance, since now you have 2 contact
databases. Making sure they are sync'd wouldn't be an automatic process
and
invariably would mean that an admin would have to fire up ADSI Edit
every
once in a while. This is mitigated by how often you change contacts. In
an
org where contacts change rarely, or never this isn't a problem. Where I
work, contacts nmber in the THOUSANDS and change EVERY DAY. The
administrative overhead of maintaining those guys in AD is brutal, and
that's why at my work I have basically banned using public folders as a
contact manager and insisted that we use SQL server with a web
front-end,
this makes things simple for the maintainer, extensible and fast, and
SQL
server plugs into everything.

hth



-----Original Message-
From: Steve Hanselman [mailto:[EMAIL PROTECTED]
Sent: Saturday, December 17, 2005 3:18 AM
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] CID lookup from an Exchange Public folder


We have a public folder full of contacts, but I understood that you
could only access this if the contacts were contacts in AD?

I was planning on doing a match on telephone number, mobile number and
fax.  And then pulling a shortened version of the name as the caller ID,

Steve


-Original Message-
From: Jason SJOBECK [mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 16 December 2005 21:48
To: asterisk-users@lists.digium.com
Cc: Steve Hanselman
Subject: Re: [Asterisk-Users] CID lookup from an Exchange Public folder

Steve,

You can get to anything in Exchange via LDAP. What is and or is not
working? Where are you entering the callerID info you want pulled?
Please see attachment for where you might want to enter this. Please
share if you get this.

Cheers.

Jason


-
Message: 1
Date: Fri, 16 Dec 2005 18:29:12 -
From: "Steve Hanselman" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] CID lookup from an Exchange Public folder
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Has anybody done this?

I looked at LDAP but you can't get to them that way, I'm considering
either a timed export, or some other way (can you access them via IMAP?
Or by wget on the owa web structure?)

Steve




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received  this communication in error, please notify Brendata i

RE: [Asterisk-Users] CID lookup from an Exchange Public folder

[Steve Hanselman]

We have a public folder full of contacts, but I understood that you
could only access this if the contacts were contacts in AD?

I was planning on doing a match on telephone number, mobile number and
fax.  And then pulling a shortened version of the name as the caller ID,

Steve


-Original Message-
From: Jason SJOBECK [mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 16 December 2005 21:48
To: asterisk-users@lists.digium.com
Cc: Steve Hanselman
Subject: Re: [Asterisk-Users] CID lookup from an Exchange Public folder

Steve,

You can get to anything in Exchange via LDAP. What is and or is not
working? Where are you entering the callerID info you want pulled?
Please see attachment for where you might want to enter this. Please
share if you get this.

Cheers.

Jason


-
Message: 1
Date: Fri, 16 Dec 2005 18:29:12 -
From: "Steve Hanselman" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] CID lookup from an Exchange Public folder
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Has anybody done this?

I looked at LDAP but you can't get to them that way, I'm considering
either a timed export, or some other way (can you access them via IMAP?
Or by wget on the owa web structure?)

Steve




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received  this communication in error, please notify Brendata immediately on:

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[Asterisk-Users] CID lookup from an Exchange Public folder

[Steve Hanselman]

Has anybody done this?

 

I looked at LDAP but you can’t get to them that way, I’m
considering either a timed export, or some other way (can you access them via
IMAP? Or by wget on the owa web structure?)

 

Steve

 






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RE: [Asterisk-Users] function cut()

[Steve Hanselman]

Added it to the Makefile, amended the
reference to cut_synopsis and it compiles, installs and works fine.

 

Logged it as a bug on mantis.

 

Steve

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 15 December 2005 16:38
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
function cut()



 

Ok, looks like app_cut is also function
cut, but is missing from the Makefile in apps?

 

If you add it in it doesn’t compile,
due to cut_synopsis not being defined.

 

Seems like it’s in a state of flux,
who’s working on this?

 

Steve

 

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 15 December 2005 14:50
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] function
cut()



 

I thought that app_cut was deprecated in favour of function
cut(), but I can’t see this in the list or the code as of
SVN-trunk-r7472M?

 

Seeing as I’ve just edited the dial plan, can anybody
shed any light on this, or should I revert back to app_cut?

 

Steve

 

 






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The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk
The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___
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RE: [Asterisk-Users] function cut()

[Steve Hanselman]

Ok, looks like app_cut is also function
cut, but is missing from the Makefile in apps?

 

If you add it in it doesn’t compile,
due to cut_synopsis not being defined.

 

Seems like it’s in a state of flux,
who’s working on this?

 

Steve

 

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 15 December 2005 14:50
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] function
cut()



 

I thought that app_cut was deprecated in favour of function
cut(), but I can’t see this in the list or the code as of
SVN-trunk-r7472M?

 

Seeing as I’ve just edited the dial plan, can anybody shed
any light on this, or should I revert back to app_cut?

 

Steve

 

 






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The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___
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[Asterisk-Users] function cut()

[Steve Hanselman]

I thought that app_cut was deprecated in favour of function
cut(), but I can’t see this in the list or the code as of SVN-trunk-r7472M?

 

Seeing as I’ve just edited the dial plan, can anybody
shed any light on this, or should I revert back to app_cut?

 

Steve

 

 






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[Asterisk-Users] Possible bug in record?

[Steve Hanselman]

I’m trying to get record to append to a file, I’m
using this:-

 

exten => 2,n,record(/tmp/${UNIQUEID}.gsm|5|0|a)

 

And it’s creating a new file?

 

If I check /tmp I can see the same filename being reused
each time, but the file jjust contains the latest recording.

 

Can anybody else confirm this?

 

Steve

 

 






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[Asterisk-Users] Sending a recorded message to voicemail

[Steve Hanselman]

Hi,

 

We have an IVR application which produces a gsm file (it’s
appended at various points, so I can’t just drop them in voicemail), I
want to send this to a users mailbox, but I can’t see a way to do this, I
presume that merely dropping the file into the directory isn’t going to
trigger off the usual notifications?

 

Any ideas?

 

Regards

 

Steve

 






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[Asterisk-Users] Goldstar GDK 186 voicemail

[Steve Hanselman]

Are there any GDK users on the list?

 

Do you know if it’s possible to disable voicemail on a
per extension basis such that it returns busy rather than diverting to
voicemail?

 

I have the manuals but I can’t see any reference to
this.

 

 

Regards

 

Steve

 

 

 






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RE: [Asterisk-Users] callfile: How to invoke SetCallerPres ?

[Steve Hanselman]

Probably easiest to set a variable to the number to be called and then
jump to an extension to do whatever you want to do?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruno
Voigt
Sent: 13 September 2005 23:37
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] callfile: How to invoke SetCallerPres ?

Hi,
how may I define in a callfile the CallerID presentation to be used for
the requested call,
eg. set it to prohibited?

TIA, Bruno


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[Asterisk-Users] Detecting retries in call files

[Steve Hanselman]

Can anybody see a way of detecting the current number of
retries remaining to a call file in the extension context that it is calling?

 

E.g. If I want to schedule a fax and I want to feed an email
back to the sender stating that the number is busy 2/5 retries remaining?

 

Steve

 

 

 






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RE: [Asterisk-Users] /etc/init.d/asterisk barfing

[Steve Hanselman]

Sorry, went of at a tangent (that's what you get for half reading emails
I guess!)

Ok, guess the easiest thing to do is to check in the contrib directory,
diff your one against the redhat (are you running redhat?)

Mine is the same and it works fine, maybe you're running an outdated
init script.

Still worth trying the strace against the service command, at least
it'll give you an idea of what it's trying to do.

One last thing, did it work and then stop working or is this a fresh
install and it's never worked?

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 01 September 2005 10:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] /etc/init.d/asterisk barfing

Hi Steve :)

The problem is not with the asterisk command, nor with safe_asterisk but

with the /etc/init.d/asterisk script

if I manually run

/etc/init.d/asterisk start

all's ok

if I manually run

service asterisk start

it says that it has started, but hasn't :)

Julian

Steve Hanselman wrote:
> Try doing an strace on it and seeing what the last section shows you.
> i.e. strace asterisk -vvvc
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Julian
> Lyndon-Smith
> Sent: 31 August 2005 22:39
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] /etc/init.d/asterisk barfing
>
> Ok, starting to get cheesed off and feeling rather silly.
>
> cvs head as of 5 minutes ago.
>
> #root asterisk -vvvc
>
> works, no problem.
>
> #root safe_asterisk
>
> works no problem
>
> #root service asterisk start
>
> Starting asterisk: [  OK  ]
>
> #root asterisk -r
> Unable to connect to remote asterisk (does /var/run/asterisk.ctl
exist?)
>
> /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after

> the run).
>
> Can't find any reasons or errors for this not working - does anyone
have
>
> any clue on where to start looking - I need * to automatically start
on
> init.
>
> Julian.
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> recipient then you are hereby notified that you have received this
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RE: [Asterisk-Users] /etc/init.d/asterisk barfing

[Steve Hanselman]

Try doing an strace on it and seeing what the last section shows you.
i.e. strace asterisk -vvvc



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 31 August 2005 22:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] /etc/init.d/asterisk barfing

Ok, starting to get cheesed off and feeling rather silly.

cvs head as of 5 minutes ago.

#root asterisk -vvvc

works, no problem.

#root safe_asterisk

works no problem

#root service asterisk start

Starting asterisk: [  OK  ]

#root asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

/var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after 
the run).

Can't find any reasons or errors for this not working - does anyone have

any clue on where to start looking - I need * to automatically start on
init.

Julian.
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[Asterisk-Users] LG Goldstar GDK-186/162 question on voicemail

[Steve Hanselman]

Are there any other GDK users out there with Asterisk?

 

I’ve got all the integration working, except
voicemail.

 

Does anybody know a way of disabling the forward to
voicemail on a per extension or per DDI basis (I can disable the voicemail hunt
group but then I can’t light the MWI indicators as it seems that only
ports marked in the voicemail group can issue the MWI on/off commands).

 

Steve

 

 






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RE: [Asterisk-Users] Re: Re: SpanDSP rxfax, no tiff

[Steve Hanselman]

Add the debug option to the rxfax line…

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Danz
Sent: 15 July 2005 13:13
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Re:
SpanDSP rxfax, no tiff



 

I’ve tried
both with and without Answer as the first line, same result.  When I was
searching through the archives, I believe there was a post from Steve Underwood
that said to always use Answer as the first line.

 

 

---

> Yes, the
permissions are okay for getting to that folder.

>
/var/spool/asterisk is writable (voicemail works & that's a subdirectory

> under the
same path that has the same permissions as the subdirectory

>
'asterisk-fax'

> 

> As the same
user that runs asterisk I did a 'touch

>
/var/spool/asterisk/asterisk-fax/test.tif' just to be sure I could write

> to that
directory.  Permissions are fine.

[...]

> [custom-fax]

> exten =>
s,1,Answer

> exten =>
s,2,Macro(faxreceive)

> exten =>
s,3,SetVar(>

> exten =>
h,1,system(/usr/local/sbin/mailfax ${FAXFILE}

>
[EMAIL PROTECTED] "${CALLERIDNUM} ${CALLERIDNAME}" ${ONZENID})

[...]

 

I do not have the
system perform an answer. Try removing the line,

perhaps the
system is getting confused.

 

 

B

 






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RE: [Asterisk-Users] Voicemail => SMS

[Steve Hanselman]

A little off topic, but I'm on orange, what's the domain and what is the format 
e.g. 07973 or +447973...



From: [EMAIL PROTECTED] on behalf of Wilson Pickett
Sent: Fri 01/07/2005 6:56
To: Mark Charlton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail => SMS



> I have been trying for a while to find a way to get an SMS send when I
> receive a voicemail into my asterisk system.  I don't want to send an
> SMS if the caller doesn't leave a message.  I have voicemail.conf set
> up to email and delete.

I use a backward solution to this problem, but it works. Orange, my
cell provider offers free SMS alerts for email sent to
[EMAIL PROTECTED] I send my vmail messages to my regular email
server which keeps them for online email retrieval. A procmail recipe
on the server then makes up an email without the vmail attachment to
my orange address with the callerid in the subject. Orange sends an
SMS that tells me I have a vmail message from ${CALLERID}. Although it
seems like a silly solution it does _exactly_ what you asked about.
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RE: [Asterisk-Users] Setting Caller ID after Dial

[Steve Hanselman]

And the UK although the PRI provider can either override or supply
it for you and you are normally limited (unless you've signed an
agreement) to DDI numbers directly provided by the PRI provider.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 30 June 2005 00:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Setting Caller ID after Dial

On 6/29/05, Bryce Chidester <[EMAIL PROTECTED]> wrote:
> The CallerID that is seen by others on calls originating from your
> PRI is set by your PRI provider; you have no control from Asterisk
> about this as it gets overridden by the provider. You must contact
> your carrier and ask them to set the CallerID for all PRI lines to
> the desired name/number.


Really? you should really do your home work before you state something
like this. In fact for most PRIs in the US you are wrong.
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RE: [Asterisk-Users] hidecallerid on analog line

[Steve Hanselman]

It depends on your telco, in the UK on an analog line we can prefix it
with 141, so in that case yes, Asterisk can do it. You to find out from
your telco whether a caller with a standard handset can do anything to
control callerid with your telco.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of chawki
hammoud
Sent: 30 June 2005 09:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] hidecallerid on analog line

In the ISDN case, setcallerid or hidecallerid can be
configured and I am aware that  Asterisk doesn't
support that on analog line. My question is whethere
there is something like add-on script or hardware that
will do the job. The teleco company provide the
callerid service, but no private number service.

--- Robert Webb <[EMAIL PROTECTED]> wrote:

>
> On Wed, 29 Jun 2005 13:56:00 -0700 (PDT)
>   chawki hammoud <[EMAIL PROTECTED]> wrote:
> > Is there a way to hide the callerid on analog line
> on
> > outgoing calls. Any ideas whether it could be done
> > through configuration, a script or hardware.
> >
> > Thanks;
> >
>
> It would have to be done through who ever provides
> your
> POTS service. They provide the caller ID to who you
> are
> calling. Some have the option to block it. Asterisk
> cannot
> be configured to do this.
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RE: [Asterisk-Users] list Searchability

[Steve Hanselman]

Might be worth asking the owner of voip-info.org if the mailing list
link can go on the left sidebar permanently?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: 28 June 2005 16:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] list Searchability

Great points Steve.  I think the best we can do is all throw the newbies
a bone ounce in a while.  Redirection to the content that is relevant is
enough to get most people on the path.  Like you said, the hardest part
is not seeing the trees for the forest.

This is the whole "teach a man to fish" parable.

It is pretty easy to tell someone
A) How to search and where to look
B) The basics of what Asterisk can do
C) How to be a good list citizen

With those tools, almost anyone can get their start here and beat the
learning curve.   Like it has been pointed out, there is not much else
we can do with so much information in a free flowing format.

Cheers,
Wiley




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of steve
szmidt
Sent: Tuesday, June 28, 2005 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

On Monday 27 June 2005 20:04, Robert Webb wrote:
> > I agree with that fact the same questions get posted, but
> > that problem is compounded by the fact the archives are not
> > really searchable. If the were as lease some users would search.
> > The archives need to be fully indexed.
>
> In a Google search box: site:lists.digium.com "What you are searching
> for"

The problem many newbies faces is TOO MUCH information. Not being able
to see
the trees because of the forest basically.

It does not matter either if it has been discussed until someone went
crazy or
died. The reason it keeps coming up is because it has not been solved.

I totally agree with why. I sure don't want to be the one babysitting
them.

These posts were simply pointing out what I think, as a former educator,
is
part of the problem. Something which is not that hard to do. And indeed
during some spare times I may put together something which is a lower
gradient for those who have a hard time getting it. I sure would like
to.

Now it could very well be that many of these people never get anywhere
because
it's just too hard for them. But I know when I started a few years back,
that
a lot of the howto's have a stiff gradient. It skips pieces of
information,
assumes knowledge which is hard to come by and so on. Standard stuff.

I'm not assuming or expecting that anyone is going to act on what I'm
saying.
If it was easy someone would have already implemented it.

But I am saying that I see there are things that CAN be done which will
make
it easier. And if it makes it easier, this list will have less "stupid"
and
repetitive questions. More people will win using Asterisk and we should
all
win. (Except those who prefer fewer people competed in this arena. And
there
are a few here who are happy it's hard for others to take part of the
fruit.
There always are.)

--

Steve Szmidt

"They that would give up essential liberty for temporary safety
deserve neither liberty nor safety."
Benjamin Franklin
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RE: [Asterisk-Users] Nasty little incident ...

[Steve Hanselman]

I doubt they do, if they are marked as being there, but happen to be down then 
the numbers would stay the same.
Sounds more likely that something happened with the clock source.

You'd need to reproduce it out of hours and look at the output of pri show span 
x and cat /proc/zaptel/*





From: [EMAIL PROTECTED] on behalf of Rich Adamson
Sent: Wed 15/06/2005 5:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nasty little incident ...



> >>We have a te410p, with the following connections:
> >>
> >>span 1 connected to a 32 Channel EuroISDN
> >>span 2 connected to a card in a legacy pbx (Meridian)
> >>span 3 connected to a 10 Channel EuroISDN
> >>span 4 connected to a card in a legacy pbx (Meridian)
> >>
> >>We have no need for the meridian now, and decided to turn it off. I did
> >>not change the zaptel.conf settings, nor the zapata.conf settings.
> >>
> >>When the meridian was turned off, * would no longer allow any outbound
> >>or inbound calls through spans 1 and 3 (although these are connected to
> >>the pstn). When I turned the meridian back on - in a hurry I might add
> >>;) (had no time to play with configurations) and restarted *, then
> >>everything was ok again ...
> >>
> >>Should I comment out span 2 and 4, run a ztcfg, unplug the cables in 2
> >>and 4, and then turn off the meridian ?
> >>
> >>Julian.
> >>
> >>/* zaptel.conf */
> >>
> >>span=1,1,0,ccs,hdb3,crc4
> >>bchan=1-15,17-31
> >>dchan=16
> >>
> >>span=2,0,0,ccs,hdb3,crc4
> >>bchan=32-46,48-62
> >>dchan=47
> >>
> >>span=3,2,0,ccs,hdb3,crc4
> >>bchan=63-77,79-93
> >>dchan=78
> >>
> >>span=4,0,0,ccs,hdb3,crc4
> >>bchan=94-108,110-124
> >>dchan=109
> >>
> >>loadzone=uk
> >>defaultzone=uk
> >>
> >>
> >
> >Just a wild guess
> >
> >When the two meridian links disappeared, the channel numbers
> >probably changed. Instead of channels 1 through 124, you probably
> >have channels 1 through 62 and your supporting dialplan (and other
> >channel specific items) likely don't match.
> >
> >
>
> I thought that the definitions in the zaptel.conf and zapata.conf (see
> below) defined the channel numbers, not the physical channels themselves
> ? I use Dial(zap/g3) to call on the zap channels.
>
> /* zapata.conf */
>
> context=isdn32-b
> prilocaldialplan=national
> internationalprefix = 00
> nationalprefix = 0
> localprefix = 01702
> group=1
> signalling=pri_cpe
> switchtype=euroisdn
> channel=1-15,17-31
>
> context=meridian-b
> group=2
> signalling=pri_net
> switchtype=euroisdn
> channel=32-46,48-62
>
> context=isdn32-a
> pridialplan=unknown
> group=3
> signalling=pri_cpe
> switchtype=euroisdn
> channel=63-77,79-93
>
> context=meridian-a
> group=4
> signalling=pri_net
> switchtype=euroisdn
> channel=94-108,110-124

I'm sure there are others on this list that can add to this, but
when the card drivers are loaded and ztfg run, the channels that
are discovered have to be mapped to what's in zaptel.conf one way or
another. (Moving card driver load around changes the discovered
order and one must manually modify zaptel.conf to match.)

Then each zap channel is defined in zapata.conf, and those definitions
have to match the channel numbers resulting from the above zaptel.conf
stuff.

So, what happens when two E1s disappear? Do the avaiable channel
numbers change at the zaptel.conf level? My best guess is they do,
but I don't have E1s around to play with to prove it. So, that's
my best guess and it certainly can be an incorrect guess on my
part.


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[Asterisk-Users] Transfers on PRI connected channel banks and legacy PBX's

[Steve Hanselman]

Hi,

 

We’re using our legacy PBX as a channel bank with
asterisk sitting between the pbx and our telco provider spliced by a TE410P.

 

If it were a straight analog FXS card then we’d use a
hook flash to break into asterisk for transfers etc, does anybody know what the
equivalent is for the PRI zaptel support?

 

 

Regards

 

Steve

 

 

Steve Hanselman

Brendata (UK) Ltd

 

Tel: +44 (0)1268 466111

Fax: +44 (0)870 1387283

Mob: +44 (0)7973 750993

 

 

 






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RE: [Asterisk-Users] Asterisk to Cisco Unity

[Steve Hanselman]

With call manager V4 and above it's extremely easy, just connect a SIP trunk to 
*.

BTW Unity is the Cisco voicemail system, Call Manager (CCM) is the actual PBX 
so your terminology may be confusing some people.




From: [EMAIL PROTECTED] on behalf of Simone
Sent: Fri 10/06/2005 10:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity



I understand what you're saying, but I am not the one who makes the
decisions. That decision is made already, so since I am actually getting
your point and I agree with that, the only thing I can try to do right
now, is try to avoid having Cisco Unity in the other 3 offices. I would
love to implement Asterisk in these ones, but if it cannot be connected
to Cisco this won't be an option at all, they won't consider it.

So, back to the question, is it possible to connect Asterisk to Cisco
and have all the functionality expected, and is it hard?

Thanks, have a nice day

Simone

William Boehlke wrote:

>By the time you install the Asterisk server you have more features than
>Cisco delivers with Unity, for half the cost and without those annoying
>viruses.
>
>So instead of thinking about connecting Asterisk, consider disconnecting
>Unity. They make excellent landfill.
>
>Regards,
>
>William Boehlke
>Signate
>
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Simone
>Sent: Thursday, June 09, 2005 9:20 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity
>
>Hi, just wondering if my question is just unusual or if it is a quite stupid
>one. Thought there would be someone having this kind of scenario, but maybe
>I'm wrong.
>
>btw, have a nice day
>
>Simone
>
>Simone wrote:
>
>
>
>>Hi all, first post. My company's office in the UK is soon going to get
>>a Cisco VoIP solution system. What I am interested in, and couldn't
>>find googling, is if it is possible to connect an Asterisk solution to
>>the Cisco system and have all the nice advantages of it (mainly
>>calling the extensions and directly reach the other office).
>>
>>Thanks, have a nice day
>>
>>Simone
>>___
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RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization

[Steve Hanselman]

Is the solution not to change the voicemail system to enable it to utilise 
other entities as the store, e.g. a pop3 server or an imap server rather than 
just flat files on disk (which should remain an option).

That way it doesn't matter where they listen to them or delete them from?

Steve




From: [EMAIL PROTECTED] on behalf of Race Vanderdecken
Sent: Sat 11/06/2005 12:52
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization



Aye, there's the rub.

"Now having said that, obviously we can't delete the message from the
local store of the POP3 client after it has been already downloaded, but
we are not talking about that, are we?"

1. Thou shall not require any brain cells on the part of the end-user.
2. Thou shall not require any settings to be set on the user's
equipment.
... More rules to follow.

Rule #3
Thou shall not require the user to delete voicemail messages
stored in their email account program by the voicemail server after they
have deleted it from their voicemail account, unless they have told the
administrator that they will do it, because the user thinks all of their
messages (voice, email, fax, paper, phone) are all stored in ROM
somewhere on the internet...

You will drive your users nuts if they can't delete it from their
message from one place. They will not understand they have to delete the
same message twice, trust me.

Race


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iassen
Hristov
Sent: Friday, June 10, 2005 7:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization

> --
>
> Message: 4
> Date: Fri, 10 Jun 2005 10:03:04 -0400
> From: David Brodbeck <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] Voicemail and MS Exchange
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>   
> Message-ID:
>
<[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"

>
> IMAP is no good.  Outlook, at least in older versions, cannot handle
both
> an IMAP account and an Exchange account at the same time.  (They can
do
> POP3 and Exchange together, though.)

Does this matter? All we are saying is that Exchange supports IMAP and
we
would use IMAP as the protocol to delete the message from the user's
mailbox. How does the user access his mailbox is his choice.

Now having said that, obviously we can't delete the message from the
local
store of the POP3 client after it has been already downloaded, but we
are
not talking about that, are we?

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RE: [Asterisk-Users] Can I hide caller id on the fly (per eachusesetting) on Bristuffed * and quadbri

[Steve Hanselman]

We do this, you need to ensure that you are allowed to control your
callerID (we had to request this from our telco)

You should then be able to use the SetCallerPres and SetCallerID to
control what (if any) number you give out.

Steve


-Original Message-
From: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: 05 May 2005 09:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can I hide caller id on the fly (per
eachusesetting) on Bristuffed * and quadbri


- Original Message -
From: "Peer Oliver Schmidt" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, May 04, 2005 11:14 PM
Subject: Re: [Asterisk-Users] Can I hide caller id on the fly (per each
usesetting) on Bristuffed * and quadbri


> Robert Rozman wrote:
>
>> I wonder if I can hide caller id for just certain users. Can I
override
>> caller id setting for show or hide on the fly from dialplan ?
>
> Did you try setcallerid("")?
> --
I tried but this will work if calling internal line. I'm after
dynamically
hiding caller id on QuadBRI outgoing ISDN calls...

I guess this is possible with settings in zapata.conf, but only per
channel - I wonder if it is possible to set this up by user or do it
from
dialplan with some command

Thanks in advance,

Rob.



> Best regards
>
> Peer Oliver Schmidt
> PGP Key ID: 0x83E1C2EA
>
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RE: [Asterisk-Users] Chan_sccp - status

[Steve Hanselman]

I think it's displaying the name of the line that the call is coming in on,
but you're expecting the name of the calling party (as I was!)

Steve


-Original Message-
From: Mark Johnson [mailto:[EMAIL PROTECTED] 
Sent: 03 May 2005 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Chan_sccp - status

Mark Johnson wrote:

> Julien Goodwin wrote:
>
>> Then why haven't you sent a backtrace? If I can see why it's crashing
>>
>> then I can fix it.
>>
>> Thanks,
>> Julien
>> chan_sccp project lead
>>  
>>
> The general consensus was that I needed to be running HEAD to make 
> this work properly.  I upraded last night to HEAD and my SCCP stuff 
> seems to working perfect!!  Thank you!!
>
> Also, I saw you are in need of a 7910 from your announcement.  If you 
> email me your shipping info offlist, I will make sure you get a couple 
> of them.
>
> Mark

While on the topic, I'm having some weird issues with the 7910's and the 
callerid.  I got them to display the outgoing calls correctly, but if I 
call from an internal SIP phone to an internal SCCP 7910, the display 
shows that that SCCP phone is calling itself until you answer.  After 
you pick up, it changes to read Unknown Number to 

Anyone have luck getting this to work?

Mark
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RE: [Asterisk-Users] Answering without ringing from PRI

[Steve Hanselman]

Have you tried the latest CVS, there was a bug relating to ALERTING which
was fixed yesterday...

-Original Message-
From: Ugur GUNCER [mailto:[EMAIL PROTECTED] 
Sent: 08 April 2005 04:54
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Answering without ringing from PRI

I made that but still same no ringing for pri coming calls  
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Mathew McKernan
> Sent: Friday, April 08, 2005 5:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Answering without ringing from PRI
> 
> Hi,
> 
> Where you have your 1st priority, I suspect you have it set 
> to "Answer".
> Try changing this to Wait(1). Then on priority 2 put answer. i.e.
> 
> Exten => s,1,Wait(1)
> Exten => s,2,Answer
> Exten => blah blah
> 
> Hope that covers it,
> 
> Thanks
> 
> Mathew
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Ugur GUNCER
> Sent: Friday, 8 April 2005 11:39 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Answering without ringing from PRI
> Importance: High
> 
> 
> 
> How can i set asterisk for when call came from pri ring once 
> then answer pri call.
> 
> In now call cames from pri then asterisk directly answering 
> pri call without ringing. Then my carries hangup call because 
> they said your box is answer without ringing 
> 
> 
> Iyi Calismalar
> Saygilarimla
> 
> 
> 
> Ugur GUNCER
> Sistem Yoneticisi
> Telebizz Tel. ve Int. Hizm. 
> 
> Office= +90 212 347 6959
> Gsm   = +90 544 535 9737
> Fax   = +90 212 347 6949
> 
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RE: [Asterisk-Users] missing ring-tone

[Steve Hanselman]

Yes, Asterisk isn't sending an
ALERTING indication on the PRI, this is our problem, sounds as though this isn't
yours though, as by the sound of it you're pure VOIP?

 

 

 

-Original Message-
From: Lars L. Christensen
[mailto:[EMAIL PROTECTED] 
Sent: 05 April 2005 17:56
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
missing ring-tone

 

Unfortunately I haven't resolved the issue yet, so at the
moment I've bypassed the Asterisk and connected the Sipura directly to
the internet.

 

I've tried to reset the Sipura with no luck. I've also
noticed I do get a ring-tone when I call from line 1 to line 2 on the sipura,
so the problem must be the Asterisk. L

 

Looking forward to hear from you...

 

Cheers, Lars

 

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 5. april 2005 18:03
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
missing ring-tone



 

We've got the same issue, I'm just starting to investigate it.

 

Have you resolved your issue now?

 

If not, I'll keep you updated on what I find.

 

Steve

 

 









From: Lars L.
Christensen [mailto:[EMAIL PROTECTED] 
Sent: 27 March 2005 16:14
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users] missing
ring-tone



 

Hi there

 

I've got a rather
irritating problem with my Asterisk server...

 

Whenever someone tries to
call me, the don't get the usual "ring-tone" when they wait for me to
pickup the phone.

 

I don't know if I've
disabled this feature somewhere in my configuration files.

 

Since I'm in Denmark,
I've got an entry in the indications.conf file pointing to Denmark
(country=dk).

 

Any ideas where to start
trouble shooting?

 

Cheers

Lars Christensen

 

 

 








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 ed information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk___
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RE: [Asterisk-Users] missing ring-tone

[Steve Hanselman]

We've got the same issue, I'm
just starting to investigate it.

 

Have you resolved your issue now?

 

If not, I'll keep you updated on
what I find.

 

Steve

 

 









From:
Lars L. Christensen [mailto:[EMAIL PROTECTED] 
Sent: 27 March 2005 16:14
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users] missing
ring-tone



 

Hi there

 

I've got a rather irritating problem with my Asterisk
server...

 

Whenever someone tries to call me, the don't get the
usual "ring-tone" when they wait for me to pickup the phone.

 

I don't know if I've disabled this feature
somewhere in my configuration files.

 

Since I'm in Denmark, I've got an entry in the
indications.conf file pointing to Denmark (country=dk).

 

Any ideas where to start trouble shooting?

 

Cheers

Lars Christensen

 

 

 






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RE: [Asterisk-Users] re: Problem: Compiling error for SpanDSP

[Steve Hanselman]

Which version of spandsp are you using, current versions check on
ASTERISK_VERSION_NUM and handle the callerid accordingly.

Some time ago the makefile changed, but yours seems fine for that as you are
carrying CCFLAGS through, ensure that you're running a recent copy (and that
you've also copied not just the spandsp but the apps as well).

Steve


-Original Message-
From: Justin Newman [mailto:[EMAIL PROTECTED] 
Sent: 30 March 2005 04:15
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] re: Problem: Compiling error for SpanDSP

> Date: Tue, 29 Mar 2005 21:43:06 -0500
> From: "KMZ Enterprises" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Problem: Compiling error for SpanDSP
>   app_rxfax
>
> After resolving my earlier problem in updating the apps Makefile with the
> patch for SpanDSP, I encountered another problem when I executed the
> "Make"
> utility from /usr/src/asterisk.  I obtained an error as shown below.  Not
> sure on how to resolve the problem.
>
>
> gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
> -Wmissing-declarations -g  -Iinclude -I../incl
> ude -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686   -DZAPTEL_OPTIMIZATIONS
> -DASTERISK_VERSION=\"CVS-HEAD-
> 03/29/05-21:11:20\" -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\"\"
> -DASTETCDIR=\"/etc/asterisk\" -D
> ASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\"
> -DASTVARRUNDIR=\"/var/run/asterisk\
> " -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
> -DASTCONFPATH=\"/etc/asterisk/
> asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\"
> -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"
> -DBUSYDETECT_MARTIN -fomit-frame-pointer  -fPIC   -c -o app_rxfax.o
> app_rxfax.c
> app_rxfax.c: In function `phase_e_handler':
> app_rxfax.c:86: structure has no member named `callerid'
> make[1]: *** [app_rxfax.o] Error 1
> make[1]: Leaving directory `/usr/src/asterisk/apps'
> make: *** [subdirs] Error 1
>
> Regards,
> Kerry

For a while, the structure switched over. If you are getting an error with
"callerid" in your build, try searching the code (app_rxfax.c) and replace
the occurances with "cid.cid_num".

Justin Newman
Newman Telecom, Inc.

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RE: [Asterisk-Users] DASS II cards supported

[Steve Hanselman]

Get a converter to Q.931, we use one called
an IQ200 from (I vaguely recall) Teltrend, search the web, works fine, easy to
setup, we've used them at two customers now with no problems at all.

 

Steve

 

 









From:
Stephen Owen hosted [mailto:[EMAIL PROTECTED] 
Sent: 08 February 2005 14:35
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DASS II
cards supported



 



I know Q931 cards are supported, does anybody know how to





go about supporting DASS II ? 





 





Thanks





 





Stephen





 





 





 








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[Asterisk-Users] RE: chan_skinny and firmware upgrade

[Steve Hanselman]

Stick that on the Wiki

-Original Message-
From: Tom Ivar Helbekkmo [mailto:[EMAIL PROTECTED] 
Sent: 24 January 2005 16:31
To: Subhi S Hashwa
Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Steve Hanselman
Subject: Re: chan_skinny and firmware upgrade

Subhi S Hashwa <[EMAIL PROTECTED]> writes:

> The xml request is a feature of new firmware, that is my guess.

Yup.  Older, skinny capable, phones request the SEP...cnf file, which
is in a binary format.  I don't know how to get them to update their
firmware, but I do know how to build the SEP file to configure the
phones correctly.  I use the below program to write a simple one, to
give the phone the name, address and TCP port number (2000) to use for
the skinny protocol (I haven't bothered to make something that reads
the configuration data from the terminal -- I edit, compile, and run):

/*
 * Program to write a SEPDefault.cnf for Cisco SCCS phones.
 * Records in the file have two bytes that tell what data
 * is supplied, then the data (either fixed length or as
 * a zero-terminated character string.
 */

#include 
#include 

int main(int argc, char **argv) {
  FILE *SEPDefault;
  unsigned char SEPData[BUFSIZ];

  SEPDefault = fopen("SEPDefault.cnf", "w");

  /* change the string to be the FQDN of the call manager: */
  sprintf(SEPData, "%c%c%s", 1, 1, "pbx.company.com");
  fwrite(SEPData, 1, strlen(SEPData) + 1, SEPDefault);
  /* change the last four numbers to the IP address of the call manager: */
  sprintf(SEPData, "%c%c%c%c%c%c", 1, 2, 192, 168, 1, 42);
  fwrite(SEPData, 1, 6, SEPDefault);
  /* the last four numbers are 2000 as a four byte little-endian integer: */
  sprintf(SEPData, "%c%c%c%c%c%c", 1, 3, 208, 7, 0, 0);
  fwrite(SEPData, 1, 6, SEPDefault);
  /* this is a special end of file marker: */
  sprintf(SEPData, "%c%c", 1, 255);
  fwrite(SEPData, 1, 2, SEPDefault);

  fclose(SEPDefault);
}

/*
 * eof
 */

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145

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RE: [Asterisk-Users] chan_skinny and firmware upgrade

[Steve Hanselman]

>From the very early days of Cisco skinny the phones have all requested
XMLDefault.cnf.xml, you just need to pop it in there (either run a tcpdump
on the tftp port or run the daemon in logging mode and you'll see).

Steve


-Original Message-
From: Subhi S Hashwa [mailto:[EMAIL PROTECTED] 
Sent: 24 January 2005 14:46
To: Steve Hanselman
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] chan_skinny and firmware upgrade

Monday, January 24, 2005, 9:23:50 AM, Steve Hanselman wrote:

> Nothing to do with skinny, drop the new file(s) in your tftp directory and
> edit the .xml file to specify the new version, the phone will upgrade
itself
> when it loads the config.

the firmware I have doesn't request xml file it requests SEP.cnf
I udnerstand the new versions of firmware request SEP.cnf.xml.

Not sure where to go from here, any ideas?





-- 
Best regards,
 Subhi S Hashwamailto:[EMAIL PROTECTED]
 When everything is heading your way, you're in the wrong lane.


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RE: [Asterisk-Users] chan_skinny and firmware upgrade

[Steve Hanselman]

Nothing to do with skinny, drop the new file(s) in your tftp directory and
edit the .xml file to specify the new version, the phone will upgrade itself
when it loads the config.

Steve


-Original Message-
From: Subhi S Hashwa [mailto:[EMAIL PROTECTED] 
Sent: 23 January 2005 06:33
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] chan_skinny and firmware upgrade

Hello all,

I am trying to upgrade the firmware on my cisco 7910 without using CCM. I
was told that
chan skinny is possibly capable of doing that and would like to make
sure.

I have P00405000600 firmware which I have put in version in
skinny.conf. the phone basiclaly stops at verifying load. tcpdump
shows nothing happening apart from small amount of traffic to port
2000 (skinny).

Does anyone have any ideas on how to get the new firmware into the
phone? cisco instructions arent very helpful.

PS> unlike the bigger brother of the phone, this one does not request
PS> OS79XX.TXT file and is not SIP capable.

  

-- 
Best regards,
 Subhi S Hashwa  mailto:[EMAIL PROTECTED]
 When everything is heading your way, you're in the wrong lane.


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RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Steve Hanselman]

Any interrupts would be useful, that's the issue, the interrupt count is
zero.

-Original Message-
From: Matt Riddell [mailto:[EMAIL PROTECTED] 
Sent: 14 January 2005 11:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

Steve Hanselman wrote:
> I'm assuming that other non Digium cards work in it, but yes, you're
right.
> 
> Has anybody run any other PCI cards in those slots under Linux and seen
> interrupts from those cards?

You'd be hard pressed to find a standard card requiring accurate 
interrupts 1000 times per second...

-- 
Cheers,

Matt Riddell
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RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Steve Hanselman]

I'm assuming that other non Digium cards work in it, but yes, you're right.

Has anybody run any other PCI cards in those slots under Linux and seen
interrupts from those cards?

Steve


-Original Message-
From: Adam Goryachev [mailto:[EMAIL PROTECTED] 
Sent: 14 January 2005 09:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

On Fri, 2005-01-14 at 09:23 +0000, Steve Hanselman wrote:
> Has anyone also logged a support call with Digium, it has to be either the
> card, Linux or the Zaptel drivers.

You missed the obvious "or the HP Compaq DL380 G4 server"

Regards,
Adam

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RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Steve Hanselman]

Has anyone also logged a support call with Digium, it has to be either the
card, Linux or the Zaptel drivers.

Steve


-Original Message-
From: Joshua McAdam [mailto:[EMAIL PROTECTED] 
Sent: 14 January 2005 06:30
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

Has anyone logged a support issue with HP on this one?

I still haven't been able to get it working so far,
So I'm going to log a support issue here in australia to see what HP can do
about this and was wondering if anyone else has.

Josh

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Monday, 10 January 2005 4:22 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

Make sure you has a span defined for each port on the TE410P. With out
signaling it would not take interrupts.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl H.
Putz
Sent: Monday, January 10, 2005 12:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4
server

I have been having this exact problem with a Tatung dual EMT-64 server
as
well.

I have been trying to get a TE410P running and all looks great, driver
loads, runs ztcfg OK, etc. but no interrupts are ever processed.

One additional piece of info that I have not seen in this thread is that
I
am able to successfully start and run a T100P card in this system.  In
the
same PCI slot, wct1xxp driver built from the same CVS HEAD version as
the
wct4xxp.

Just hoping this might shed some light on the problem for any Digium
folks
monitoring the forum.


Karl Putz



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RE: [Asterisk-Users] fax to email

[Steve Hanselman]

Has anybody looked into implementing a fax send interface for Asterisk using
the FSP code, that way it would plug straight into outlook and all the other
windows bits'n'pieces?



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RE: [Asterisk-Users] phones with two ethernet ports

[Steve Hanselman]

Some of the Cisco phones do (7940,7960 etc, also the 7910+SW but this is
skinny only).

Steve


-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED] 
Sent: 02 January 2005 21:35
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] phones with two ethernet ports

Hi there, what phones are available that have two ethernet ports?
I want to do some cabling at a new installation and i heard there are
such phones (SIP i guess) out there. That way i dont have to run two
cat5 to the user desktop.
I think 3COM had one but can't find the web site reference for the two
port phone

thanks,

erick
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RE: [Asterisk-Users] Final call for departments

[Steve Hanselman]

"Accounts" by itself would be useful.


-Original Message-
From: David Boyd [mailto:[EMAIL PROTECTED] 
Sent: 30 December 2004 00:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Final call for departments

HOw about :

development


Dave
On Wed, 2004-12-29 at 04:51, Alspach Family wrote:
> I am getting ready to submit a list of department names to be recorded.  
> This is what I have so far:
> 
> Accounting
> Accounts payable
> Accounts receivable
> Administration
> Billing & Collections
> Complaint
> Customer Service
> Engineering
> Facilities
> Help desk
> Human Resources
> Information Technology
> Inside Sales
> Investor Relations
> Legal
> Mail room
> Marketing
> Printing
> Projects
> Public Relations
> Purchasing
> Receiving
> Sales
> Sales Floor
> Shipping
> Shop
> Support
> Systems
> Technical Support
> Travel
> 
> If any one has additional suggestions, please e-mail them to me 
> ([EMAIL PROTECTED] or [EMAIL PROTECTED]).  I am fairly sure that 
> none of the above exist (I was only able to search through the WIKI 
> list, so if there are other prompts in the CVS that are not listed 
> there, I do not know about them.)  If I have made a dupe, please let me 
> know so that I can remove it.  I was fairly certain that 'Operator' was 
> already available but I was unable to find it by its self. 
> Thanks for your help.
> I plan on sending these off on Friday the 31st so please try to get them 
> to me by then.
> 
> Thanks;
> James
> 
> 
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RE: [Asterisk-Users] Music instead of Tunes

[Steve Hanselman]

Thinking about it we may well be able to do this in the UK as one of the
complaints I get about Asterisk is that our ring tone has changed to
external callers, (the zone is set correctly for zaptel, but it's different
from the normal ring tone), so the tones must be coming from the TE405, not
just generated as a result of the accept (unless some data in the accept
signifies the tones to generate?)



-Original Message-
From: Paul Crick [mailto:[EMAIL PROTECTED] 
Sent: 29 December 2004 18:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Music instead of Tunes

> So we could provide caller position announcements
> without the callers actually incurring charges?
> Has anybody tried this (in the UK)?
Maybe.. but probably not..

In the UK (and most European countries), the ACCEPT message triggers
generation of ringback tone at the calling party's exchange (central
office). This is as opposed to the North American way of doing things where
the ACCEPT message opens up a one way speech path from the called party to
the calling party (originally for providing inband call progress tones I
believe). Also, there's a timer on how long you can be in that state
without issuing an ANSWER and thus tripping answer supervision/billing
commencement.

I think technically it IS possible to get UK kit to work in the US fashion,
but you have to talk to a switch tech that knows what he's doing, and of
course you may get bitten with the "Yeah, it's doable, but we don't have
that software feature pack installed on our switch" line.

Paul

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RE: [Asterisk-Users] Music instead of Tunes

[Steve Hanselman]

So we could provide caller position announcements without the callers
actually incurring charges?

Has anybody tried this (in the UK)?

-Original Message-
From: Peter Svensson [mailto:[EMAIL PROTECTED] 
Sent: 29 December 2004 14:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Music instead of Tunes

On Wed, 29 Dec 2004, Steve Hanselman wrote:

> > Are the two cases different in any way? The external call comes in, goes

> > to a context which eventually leads to a Dial(...) calling the internal 
> > user. That Dial call provides music to the external caller while the 
> > internal call is in progress.

> The difference is that you'd have to answer the call, my guess is that it
> can't be done (by a Joe Average like ourselves), otherwise we'd provide
> useful information to callers at no charge.

For pots lines this is true.

For isdn lines there is no need to answer prior to sending data. The 
reverse path (from the called party towards the calling party) is opened 
when (this is form memory, it may be another IE) PROGRESS is transmitted. 

You can use Playback and a host of other connads on an unanswered line. 
Some of these will automatically answer the line unless given an option 
not to.

Peter


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RE: [Asterisk-Users] Music instead of Tunes

[Steve Hanselman]

The difference is that you'd have to answer the call, my guess is that it
can't be done (by a Joe Average like ourselves), otherwise we'd provide
useful information to callers at no charge.


-Original Message-
From: Peter Svensson [mailto:[EMAIL PROTECTED] 
Sent: 29 December 2004 11:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Music instead of Tunes

On Wed, 29 Dec 2004, Steve Hanselman wrote:

> > On Tue, 28 Dec 2004, Marc Storck wrote:
> > 
> > > more and more operators in Europe offer music instead of ring tunes. 
> > > E.g. instead of the 400 Hz or whatever tunes, the caller will hear
J-Lo, 
> > > or Mozart Currently I will have to answer the line to do that. Is 
> > > there a way to do this with asterisk?
> > 
> > See the help for dial:
> >'m' -- provide hold music to the calling party until answered.
> 
> I'm guessing he wants to do it the other way around, i.e. the external
> calling party hears music, not the internal calling party making an
external
> call.

Are the two cases different in any way? The external call comes in, goes 
to a context which eventually leads to a Dial(...) calling the internal 
user. That Dial call provides music to the external caller while the 
internal call is in progress.

Asterisk has no concept of external or internal callers, only channles and 
contexts.

Peter


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RE: [Asterisk-Users] Music instead of Tunes

[Steve Hanselman]

I'm guessing he wants to do it the other way around, i.e. the external
calling party hears music, not the internal calling party making an external
call.


-Original Message-
From: Peter Svensson [mailto:[EMAIL PROTECTED] 
Sent: 28 December 2004 21:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Music instead of Tunes

On Tue, 28 Dec 2004, Marc Storck wrote:

> more and more operators in Europe offer music instead of ring tunes. 
> E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, 
> or Mozart Currently I will have to answer the line to do that. Is 
> there a way to do this with asterisk?

See the help for dial:
   'm' -- provide hold music to the calling party until answered.

Peter


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RE: [Asterisk-Users] Echo on one E1 line, but not the other

[Steve Hanselman]

More by trial and error, we backed off the gain until it disappeared but
with no detriment to the call quality (didn't want people to sound like a
whisper).

Our situation was somewhat different to yours though, we were seeing the
issues on calls from our PBX, not on calls through the IP phones.

Who hears the echo, the IP phone user or the remote user?

Steve


-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED] 
Sent: 13 December 2004 12:36
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other

Both ISDN lines are going into the same * box - span 1 is the "test" isdn
line and span 3 is the "live" isdn line. The two ISDN lines are situated
right next to each other!

As mentioned there is no problem with the test line, so there isn't a
problem with * as such (I don't think!). Perhaps I haven't got the
configuration quite right. 

When you say that you adjusted the gain, is that the tx/rx settings in
zapata.conf ? How did you determine the correct settings: by placing a call
and monitoring using ztmonitor ?

Thanks.

Julian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Hanselman
Sent: 13 December 2004 12:13
To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other

It looks like this is a splice between a couple of ISDN-30 lines and one or
more PBX's?

Are they both with the same provider, or with different providers?

We ended up adjusting the gain our ours as we would hear a distinct echo on
certain calls.

Other than that, you'll need to do the usual tests, check for shared
interrupts, also, see if disk activity causes a problem.

(check /proc/interrupts for shared interrupts).

Have you also checked the output of the pri commands to ensure that you're
not getting line errors?

Steve


-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED]
Sent: 13 December 2004 12:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo on one E1 line, but not the other

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.

We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.

During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky me. However, since we
have started rolling out, we've had quite loud complaints that there is "a
terrible echo". If I direct my 7960 to use the primary line, there is an
echo. If I use the second line (dialling the same external number) there is
no echo at all.

What could be the issue ? I have noticed on the primary line there is a
"detected rx/tx on channel xx, echo cancellation disabled" (or something
like that. Is this the cause ? We have several fax / modems going through
the line - should I always dedicate a channel to them ?

I've included my zaptel.conf and zapata.conf file below. Any help / comments
appreciated.

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg #

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

loadzone=uk
defaultzone=uk

#

;zapata.conf

[general]

[trunkgroups]

[channels]
musiconhold=default
language=en
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callerid=asreceived
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
;callgroup=1
;pickupgroup=1
;adsi=yes

context=isdn32-b
pridialplan=unknown
group=1
signalling=pri_cpe
switchtype=euroisdn
channel=1-15,17-31

context=meridian-b
group=2
signalling=pri_net
switchtype=euroisdn
channel=32-46,48-62

context=isdn32-a
pridialplan=unknown
group=3
signalling=pri_cpe
switchtype=euroisdn
channel=63-77,79-93

context=meridian-a
group=4
signalling=pri_net
switchtype=euroisdn
channel=94-108,110-124

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RE: [Asterisk-Users] Echo on one E1 line, but not the other

[Steve Hanselman]

It looks like this is a splice between a couple of ISDN-30 lines and one or
more PBX's?

Are they both with the same provider, or with different providers?

We ended up adjusting the gain our ours as we would hear a distinct echo on
certain calls.

Other than that, you'll need to do the usual tests, check for shared
interrupts, also, see if disk activity causes a problem.

(check /proc/interrupts for shared interrupts).

Have you also checked the output of the pri commands to ensure that you're
not getting line errors?

Steve


-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED] 
Sent: 13 December 2004 12:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo on one E1 line, but not the other

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.

We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.

During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky me. However, since we
have started rolling out, we've had quite loud complaints that there is "a
terrible echo". If I direct my 7960 to use the primary line, there is an
echo. If I use the second line (dialling the same external number) there is
no echo at all.

What could be the issue ? I have noticed on the primary line there is a
"detected rx/tx on channel xx, echo cancellation disabled" (or something
like that. Is this the cause ? We have several fax / modems going through
the line - should I always dedicate a channel to them ?

I've included my zaptel.conf and zapata.conf file below. Any help / comments
appreciated.

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

loadzone=uk
defaultzone=uk

#

;zapata.conf

[general]

[trunkgroups]

[channels]
musiconhold=default
language=en
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callerid=asreceived
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
;callgroup=1
;pickupgroup=1
;adsi=yes

context=isdn32-b
pridialplan=unknown
group=1
signalling=pri_cpe
switchtype=euroisdn
channel=1-15,17-31

context=meridian-b
group=2
signalling=pri_net
switchtype=euroisdn
channel=32-46,48-62

context=isdn32-a
pridialplan=unknown
group=3
signalling=pri_cpe
switchtype=euroisdn
channel=63-77,79-93

context=meridian-a
group=4
signalling=pri_net
switchtype=euroisdn
channel=94-108,110-124

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RE: [Asterisk-Users] XML to monitor queues on Cisco display ?

[Steve Hanselman]

You can have a refresh interval on the XML though which achieves the same
thing.

Also, you can do a push, there are examples in the developers kit available
on the Cisco web site.

-Original Message-
From: Wayne Sheppard [mailto:[EMAIL PROTECTED] 
Sent: 04 December 2004 18:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] XML to monitor queues on Cisco display ?

Henry Devito wrote:

>I attempted this but I got stuck on one issue.  Cisco phones pull data so I
>couldn't get them to autoupdate. In other words push data to them.  I am
>working on an app to run on a windows desktop that will show the queues,
the
>amount of calls in each queue, the longest wait time and the average wait
>time. I am also planning on creating the app with alarm thresholds.  When
>the app is minimized it will go to the task bar and if the queue gets too
>full it will popup the window on the desktop and/or make the icon in the
>taskbar turn red.  
>
>Henry
>  
>
Henry, that is a very useful app indeed! Do you plan to share that, sell 
it, ??
Love to get more info or help..

Cheers,
Wayne
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RE: [Asterisk-Users] Remote Office question, Draytek , recommende d analog phone

[Steve Hanselman]

I'd back that comment, we've about 6 2600VG's, the wireless drops
frequently, it's not too hot on the voice side and the VPN's drop regularly,
or did until RC4, now they're stable but the box locks up once or twice a
day.

We're working to get them sorted out, but it looks as though they'll be
RMA'd fairly shortly.


-Original Message-
From: Voip Business [mailto:[EMAIL PROTECTED] 
Sent: 02 November 2004 19:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Remote Office question, Draytek , recommended
analog phone

Shawn...

I have too many problems with a customer with a Vigor2600V ( I have
sell something about 6) and all of them are not good (they looses
registration for some reazon)

Note: 

this happens with Asterisk, SER proxy, Snom Proxy and a Voicemaster SIP

Conclussion: NOT RECOMENDED


But the vigor 2300 and others are a reall good stuff for "security"
and VPN are really great.

By the way has someone in here connect a Sipura behind a Vigor 2200 -
2300 BB router?

Regards


HA

On Tue, 2 Nov 2004 10:08:20 -0700, Shawn Dillon <[EMAIL PROTECTED]>
wrote:
> 
> 
> 
> If there a way to connect an IP phone at a remote office to a device (like
> the Sipura 3000) that would allow local POTS failover in case of the
> Asterisk or VPN going down?
> 
>  
> 
> I know the Sipura would allow a analog phone to connect to the local POTS
in
> case of a * failure, we just want to standardize on IP phones ( Polycom
600)
> all around.
> 
>  
> 
> Also has anyone had experience with the Draytek VOIP Wireless routers?
> 
>  
> 
> And finally, if we need to use a Sipura 3000 in the remote offices is
there
> any benefit with going with a Sayson analog phone versus any other?
> 
>  
> 
> Thanks
> 
>  
> 
> Shawn Dillon
> 
>  
> ___
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RE: [Asterisk-Users] Grandstream phone price

[Steve Hanselman]

You multiply to get the dollar price.

Careful where you go on holiday, it could be costing more than you think!!

-Original Message-
From: David J Carter [mailto:[EMAIL PROTECTED]
Sent: 11 October 2004 08:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream phone price

$1.64 to the £1 I think this morning so $35 stands.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wolf N.
Paul
Sent: 11 October 2004 07:40
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream phone price


Except that £55 is more like $75-80 and not $35.

Regards, Wolf


"David J Carter" <[EMAIL PROTECTED]> writes:

> I beleive Telappliant in the UK are doing them for £55, ($35)
>
> http://www.voiptalk.org/products/index.php?cPath=27
>
> Dave
>
> Grandstreams are availabe for $65 quanity one, so its not hard to believe
> that you could get them
> for $55 for larger quantities
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RE: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

[Steve Hanselman]

 
check your musiconhold.conf, for each one you define you'l get an instance.


-Original Message-
From: Matthew Boehm
To: [EMAIL PROTECTED]
Sent: 03/09/04 15:04
Subject: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

Is there any reason why there should ever be more than 1 instance of
mpg123
running on a * server?

I just did an 'uptime' and noticed all 3 of my loads where over 3.00.

'top' showed 8 mpg123 processes all processing the same 3 songs (our
background music).

I tried to kill one of them but another one spawned in its place.

Any ideas?

Thanks,
Matthew

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RE: [Asterisk-Users] Closing bug reports without fixing the repor ted problem

[Steve Hanselman]

 
 [vested interest: I reported that bug!] 

I agree with this, a system that has x bugs still has x bugs even if you
close them, I think they should be closed by houskeeping if they can't be
reproduced and the person that reported them can't provide a way to
replicate, but they shouldn't be closed because nobody has fixed them for a
while.

This seems to be the case on a number of entries. 

Steve 

 

-Original Message-
From: Rob Wise
To: [EMAIL PROTECTED]
Sent: 25/08/04 07:01
Subject: [Asterisk-Users] Closing bug reports without fixing the reported
problem

While this is more a question for the bug system admins, it does
impact on the whole Asterisk community so I'm going to ask it here.

Why does it seem to be normal procedure to close bug reports which
have not been resolved?  Twice now bug 1915 has been closed off
without a solution being found, much to the protest of those who are
suffering from the reported problem.  I am curious to know why this is
being done as it does not seem beneficial in any way to the Asterisk
project to hide problems by closing the ticket rather than leaving
them open so they can be worked on.

I also am curious to know what other members of the community think of
this.

Rob

Disclaimer: I am one of those affected by the bug, and it is still
present in today's CVS head.
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RE: [Asterisk-Users] App.c

[Steve Hanselman]

Delete it and cvs update will retrieve it.


-Original Message-
From: AJ Grinnell [mailto:[EMAIL PROTECTED] 
Sent: 02 August 2004 17:33
To: Asterisk
Subject: [Asterisk-Users] App.c

Can someone tell me where I can get just app.c from. Mine somehow got
corrupted, and no updates or anything else will fix it. I just need the one
file from the latest cvs. 8-1-04. Please help


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RE: [Asterisk-Users] TE405P and E1

[Steve Hanselman]

Our ISDN 30 is delivered that way, but we were also supplied with a balun
that takes the two balanced coaxs and turns them into a single RJ45, maybe
your telco needs to supply you with some extra kit?

Steve


-Original Message-
From: Kim Esben Jørgensen [mailto:[EMAIL PROTECTED]
Sent: 26 July 2004 10:40
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] TE405P and E1

Hello

Im from Denmark and i've just got my Digium TE405P. But i have some
problems when i connect it to my E1 connection (ISDN30).
My telco delivered a alcatel box witch have a G.703 120ohm (DB9 with a
serial to rj45) and a 75ohm coax connection.
I've tried to connect using the 120ohm with rj45 and a ordinary utp cable.
But it dosent seem to work. I've tried several zaptel.conf setting, but
none of them made the led stop blinking.


span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
loadzone=us
defaultzone=us


Do you have any hints.

regards
Kim Esben Joergensen
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RE: [Asterisk-Users] Daytime - Nighttime

[Steve Hanselman]

Yes, you'd have a dialplan entry that set a value in the database, then
acted upon that.

You'd probably want some nice voice prompts

"The system is currently in [Day/Night/Holiday] mode, press 1 to set to day,
2 to set."

Steve


-Original Message-
From: Massimo De Nadal [mailto:[EMAIL PROTECTED] 
Sent: 22 July 2004 13:57
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Daytime - Nighttime

Is it possible to build a dialplan in which shifting from daytime to
nightime is not hour based but phone driven ???


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RE: [Asterisk-Users] Caller based routing

[Steve Hanselman]

In your dialplan for your voip routing you'd put a gotoif that jumped to
your PSTN context if it matched your criteria (e.g. EXTEN = faxextension)

Steve


-Original Message-
From: GIBERT Frédéric
To: [EMAIL PROTECTED]
Sent: 21/07/04 13:58
Subject: [Asterisk-Users] Caller based routing

Hello,

Can someone explain me how to do caller based routing.

Here is my example.

I have an asterisk between a PBX and the PSTN. The second company get
the same, and so, I can interconnect them by VoIP. Classic architecture,

My problem is when I want to place fax.

The calls between the 2 sites are in gsm codec. So the fax doesn't work!

Is there any possibilities to do caller based routing in asterisk, in
order that when a fax try to send a fax, the call is automatically
routed through the PSTN and not through the VoIP.

Thanks.


GIBERT Frédéric

Mobile: +33 6 72 08 35 16

Fax : +33 1 30 71 39 33

Mail :  
[EMAIL PROTECTED]


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RE: [Asterisk-Users] Adding voice mail box

[Steve Hanselman]

They only get created as they are used and voicemail left, try leaving a
message and you should see that the structure etc is created.

Steve


-Original Message-
From: Steve [mailto:[EMAIL PROTECTED] 
Sent: 19 July 2004 08:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Adding voice mail box

On Monday 19 July 2004 01:23 am, Brian K. West wrote:
> Dont have to.. just add it to the  voicemail.conf and it will auto do
> everything for you.
>
> bkw

Well, after having restarted * a few times, and rebooted once, I can say
that 
no mailboxes were created automatically. I'm running a week old HEAD.

Brian, what version were you running when you observed this nice feature?

> - Original Message -
> From: "Steve" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, July 18, 2004 9:58 PM
> Subject: [Asterisk-Users] Adding voice mail box
>
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA1
> >
> > Hi,
> >
> > I've forgotten the command to add a vm box, and searching google and
wiki
>
> I'm
>
> > surpriced I cannot find it. I'd love to know where this is written, so I
>
> can
>
> > see how I managed to miss it!
> >
> > - --
> > Steve
> >
> > "They that would give up essential liberty for temporary safety deserve
> > neither liberty nor safety."
> > Benjamin Franklin
> >
> > -BEGIN PGP SIGNATURE-
> > Version: GnuPG v1.2.4 (GNU/Linux)
> >
> > iD8DBQFA+zjhljK16xgETzkRAh8jAKCJ7iJhFBVRxBFzbl8cGziqbnUjoQCdEzbb
> > oTA7sXW1EXmmDGpUXrPf174=
> > =zANK
> > -END PGP SIGNATURE-
> > ___
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-- 
Steve

"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
Benjamin Franklin

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RE: [Asterisk-Users] ACD Issues

[Steve Hanselman]

Sorry, my mistake, I thought you'd said you had assigned priorities, let me
go back and search for your original posting

Steve


-Original Message-
From: Robert Jackson [mailto:[EMAIL PROTECTED] 
Sent: 16 July 2004 14:10
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ACD Issues

That would certainly make sense, but I am not sure how to set an Agent's
priority.  The only information that I have been able to find is setting
a QUEUE_PRIO value when queuing the calls (New as of July 2004).

Thanks,

Robert Jackson

> -Original Message-
> From: Steve Hanselman [mailto:[EMAIL PROTECTED] 
> Sent: Friday, July 16, 2004 3:59 AM
> To: '[EMAIL PROTECTED]'
> Subject: RE: [Asterisk-Users] ACD Issues
> 
> 
> Your priority assignment will probably cause that, if you 
> have 2 people at different priorities asterisk will only send 
> to the other priority if the best 1 is busy.
> 
> I'm guessing you really want a %age split?  Whereby a is 
> guaranteed 70% of the calls and b 30%?
> 
> We used to achieve this on our old system by putting people 
> into the queues multiple times and then having it round robin.
> 
> We use the priorities to allow other agents to service calls 
> rather than just leaving people hanging.  For instance our 
> network sales team can also service the server sales queue 
> but at a lower priority, they only get the calls if all the 
> agents in the server sales queue are busy.
> 
> Steve
> 
> 
> -Original Message-
> From: Robert Jackson [mailto:[EMAIL PROTECTED] 
> Sent: 14 July 2004 23:08
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] ACD Issues
> 
> That worked great!  Thanks for the help.  Any ideas on the 
> uneven distribution problems?  Right now the agent with the 
> lowest agent number is getting 45% of the calls.  She is 
> going crazy!  
> 
> Just trying to figure out what I screwed up.
> 
> Thanks,
> 
> Robert Jackson
> Pro-Medical, Inc.
> 
> > -Original Message-
> > From: Chris A. Icide [mailto:[EMAIL PROTECTED]
> > Sent: Wednesday, July 14, 2004 12:56 PM
> > To: Robert Jackson
> > Subject: Re: [Asterisk-Users] ACD Issues
> > 
> > 
> > On 05:01 AM 7/14/2004, Robert Jackson wrote:
> >  >1) Our agents for the main call center are responsible to
> > make calls  >when they have not already received an ACD call. 
> >  However it seems that  >if they make an outbound call 
> > asterisk is still routing inbound calls to  >them.  The ACD 
> > call beeps at them via the call waiting features then if  
> > >the agent does not answer the ACD call it logs the agent
> > out.  I am just  >trying to figure out how I can tell the
> > system that the extension is  >busy.  Should I be using the 
> > new replacements to incominglimit?
> > 
> > In your cisco phones, set call waiting to off.  This way when
> > you have a 
> > phone call in progress, asterisk will be aware your phone is 
> > busy and won't 
> > send the call, versus, sending a call, and not having the 
> > agent answer it.
> > 
> > Otherwise, you have to use incoming limit because there is no
> > way to use 
> > set/checkgroup with agents as they aren't handled as devices 
> > you can attach 
> > a setgroup/checkgroup to.
> > 
> > 
> > Chris A. Icide
> > 332 Valdez Ave.
> > Half Moon Bay, CA 94019
> > 650-712-8223 voice
> > 212-400-1698 IP voice
> > 650-712-8995 fax
> > 
> > 
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RE: [Asterisk-Users] ACD Issues

[Steve Hanselman]

Your priority assignment will probably cause that, if you have 2 people at
different priorities asterisk will only send to the other priority if the
best 1 is busy.

I'm guessing you really want a %age split?  Whereby a is guaranteed 70% of
the calls and b 30%?

We used to achieve this on our old system by putting people into the queues
multiple times and then having it round robin.

We use the priorities to allow other agents to service calls rather than
just leaving people hanging.  For instance our network sales team can also
service the server sales queue but at a lower priority, they only get the
calls if all the agents in the server sales queue are busy.

Steve


-Original Message-
From: Robert Jackson [mailto:[EMAIL PROTECTED] 
Sent: 14 July 2004 23:08
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ACD Issues

That worked great!  Thanks for the help.  Any ideas on the uneven
distribution problems?  Right now the agent with the lowest agent number
is getting 45% of the calls.  She is going crazy!  

Just trying to figure out what I screwed up.

Thanks,

Robert Jackson
Pro-Medical, Inc.

> -Original Message-
> From: Chris A. Icide [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, July 14, 2004 12:56 PM
> To: Robert Jackson
> Subject: Re: [Asterisk-Users] ACD Issues
> 
> 
> On 05:01 AM 7/14/2004, Robert Jackson wrote:
>  >1) Our agents for the main call center are responsible to 
> make calls  >when they have not already received an ACD call. 
>  However it seems that  >if they make an outbound call 
> asterisk is still routing inbound calls to  >them.  The ACD 
> call beeps at them via the call waiting features then if  
> >the agent does not answer the ACD call it logs the agent 
> out.  I am just  >trying to figure out how I can tell the 
> system that the extension is  >busy.  Should I be using the 
> new replacements to incominglimit?
> 
> In your cisco phones, set call waiting to off.  This way when 
> you have a 
> phone call in progress, asterisk will be aware your phone is 
> busy and won't 
> send the call, versus, sending a call, and not having the 
> agent answer it.
> 
> Otherwise, you have to use incoming limit because there is no 
> way to use 
> set/checkgroup with agents as they aren't handled as devices 
> you can attach 
> a setgroup/checkgroup to.
> 
> 
> Chris A. Icide
> 332 Valdez Ave.
> Half Moon Bay, CA 94019
> 650-712-8223 voice
> 212-400-1698 IP voice
> 650-712-8995 fax 
> 
> 
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RE: [Asterisk-Users] Flag Bad PRI Channel

[Steve Hanselman]

You can shut down the span in its entirety, or just exclude some channels in
zaptel.conf.

Steve


-Original Message-
From: Shawn Lawrence [mailto:[EMAIL PROTECTED] 
Sent: 16 July 2004 08:03
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Flag Bad PRI Channel

Is there a way to flag a bad or noisy PRI channel in Asterisk so it will
be skipped over?

Shawn

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RE: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode

[Steve Hanselman]

On the /proc/zaptel it was lost interrupts, you haven't got any so that's
good!

On opencall.org there's a known issues link and that mentions some fax
machines that have issues, might be worth a quick check there.

I can receive from our fax machine this end, if you'd like I'll send you a
test fax if you send me your fax number, if you can receive from us and tpc
then it's more likely that the issue is with your fax machine and you may
have to wait for Steve to release his latest version.  You'll then have the
choice of rxfax or hylafax.

One last question, how is your fax machine connected, direct to a Telco pots
line or via an fxo card in asterisk?

Steve



-Original Message-
From: Stephen J. Wilcox [mailto:[EMAIL PROTECTED] 
Sent: 14 July 2004 15:43
To: Steve Hanselman
Cc: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spa
ndsp fails to decode

Hi Steve,
 not boring at all, I'm out of ideas and I'm not clued up about how to look
into
this problem further so your suggestion is appreciated.

Looking into /proc/zaptel/1 what am I looking for?
# more /proc/zaptel/1 
Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS/CRC4
ClockSource 
   1 WCT1/0/1 ClearChannel (In use) 

  16 WCT1/0/16 HDLCFCS (In use) 
  17 WCT1/0/17 ClearChannel 


Listening to audio isnt something that I think I will be able to diagnose,
I'll 
give it a go but it'll sound like a fax machine to me ;) [now, give me v90
any 
day and i'll tell you whats going on! ;) ]

I'm testing from the office fax machine which is used a lot all the time and

never has problems, I tried the tpc.int and the fax comes thro fine.

Hmm so what is that telling me? The office fax really does send dozens of
faxes 
per day with no failures

Looking at the debug output of the two, heres the ey differences:

"Slow carrier up
Slow carrier down" - prior to start receiving, altho slow carrier up is
shown in 
both a couple times once transmission begins.

The good one goes into 'start rx document' but the bad one seems to keep
trying 
to train.. i have put a diff in below .. is there anything useful here?

Thanks for any help!

Steve

# diff fax-good-tpc.int fax-bad-office 
1c1
< -- Executing RxFAX("Zap/1-1", 
"/var/spool/asterisk/faxes/20040714-140202.tif") in new stack
---
> -- Executing RxFAX("Zap/2-1", 
"/var/spool/asterisk/faxes/20040714-143316.tif") in new stack
2a3,4
> Slow carrier up
> Slow carrier down
22c24
< <<< TSI: 43 74 65 6e 72 65 74 6e 49 20 6e 6f 6d 65 44 20 20 20 20 20 20
---
> <<< TSI: 43 31 37 31 31 36 35 34 35 34 38 30 20 20 20 20 20 20 20 20 20
24,25c26,27
< Remote fax gave TSI as: "Demon Internet"
< <<< DCS: 83 00 46 f0 00
---
> Remote fax gave TSI as: "08454561171"
> <<< DCS: 83 00 86 90 00
31c33
< R8x7.7lines/mm and/or 200x200pels/25.4mm OK
---
> 2D coding OK
34c36
< Minimum scan line time: 0ms
---
> Minimum scan line time: 5ms
39a42,49
> Coarse carrier frequency 1698.91 (60)
> Training error 93.312809
> Training succeeded (constellation mismatch 48.327239)
> Fast carrier trained
> Fast carrier down
> Trainability test failed - longest run of zeros was 23
> >>> FTT: 44
> Fast carrier up
44,46c54,56
< Coarse carrier frequency 1700.20 (56)
< Training error 5.448260
< Training succeeded (constellation mismatch 13.584253)
---
> Coarse carrier frequency 1699.55 (60)
> Training error 61.497972
> Training succeeded (constellation mismatch 40.535409)
49,55c59,60
< Changed from phase 5 to 4
< Start rx document - compression 1
< Start rx page
< >>> CFR: 84
< HDLC underflow in state 5
< Post trainability
< Changed from phase 4 to 5
---
> Trainability test failed - longest run of zeros was 31
> >>> FTT: 44
61,63c66,68
< Coarse carrier frequency 1700.15 (56)
< Training error 5.155357
< Training succeeded (constellation mismatch 11.403019)
---
> Coarse carrier frequency 1700.09 (60)
> Training error 46.800474
> Training succeeded (constellation mismatch 33.977520)
65,71d69
< Fax3Decode1D: Warning, (FakeInput): Premature EOL at scanline 2155 (got 0,

expected 1728).
< Page 1 of /var/spool/asterisk/faxes/20040714-140202.tif:
< 2156 rows received
< 0 total bad rows
< 0 max consecutive bad rows
< Rx page end detected
< Changed from phase 5 to 3
73,81c71,72
< Slow carrier up
< <<< MPS: 4f
< MPS with final frame tag
< In state 5
< Changed from phase 3 to 4
< Start rx page
< >>> MCF: 8c
< HDLC underflow in state 7
< Changed from phase 4 to 5
---
> Trainability test failed - longest run of zeros was 77
> >>> FTT: 44
87,113c78,80
< Coarse

RE: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode

[Steve Hanselman]

Sorry to bore you more with the clock issue, but have you check
/proc/zaptel/ to make sure it's not missing interrupts?

There's also an option to record the audio for the fax, you could listen to
that vs a recorded file that will receive correctly on a fax machine and see
whether there is an obvious difference? (Good luck, that'll be really
scraping the barrel!!)

Does it matter where you're faxing from?  (I'm wondering whether there's an
issue with a specific machine? (You can use http://www.tpc.int to send test
faxes, you'll get some extra info if a fax fails as to why in the
transmission report).

Can you successfully send faxes out from your system, what do they look like
at the remote end?

Steve


-Original Message-
From: Stephen J. Wilcox [mailto:[EMAIL PROTECTED] 
Sent: 13 July 2004 18:12
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spandsp
fails to decode

Okay having taken in some suggestions and googled this topic to death I'm
still 
stuck - anyone got any ideas?

To recap, the faxes are coming in via a digium E1 card but failing to train 
properly or if they manage it sending a garbled and very truncated fax.

A number of folks have suggested clock sync issues.. my zaptel.conf is set
to 
use the PRI as primary clock, i have no evidence of issues altho dont know
how 
to check (other than the call quality is fine, no clicks, no pri down/ups).

What can i try?

Steve

On Mon, 12 Jul 2004, Stephen J. Wilcox wrote:

> Hi,
>  I just sent this to Steve Underwood, but then found a bunch of posts on
the
> mailing list about similar issues.. does anyone have the fix?
> 
> I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif
3.5.7
> 
> one thing i just noticed is that calls come in with format '72' which is
> G711A-law or LinearPCM.. it uses PCM for the call, i assume this is ok
> 
> the results of RxFAX vary, it sometimes saves the file in which case i get

> errors: 
> Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 0
(got 
> 2383, expected 1728).
> Fax3Decode2D: (FakeInput): Bad code word at scanline 1 (x 137).
> 
> and the resulting tif looks to be only a few rows long
> 
> or more commonly it just fails entirely.. i paste the output below so you
can 
> see. is there anything obvious i'm doign wrong here?
> 
> TIA! Steve.
> 
> -- Executing RxFAX("Zap/1-1", 
> "/var/spool/asterisk/faxes/20040712-183339.tif") in new stack
> Changed from phase 0 to 1
> Start receiving document
> Changed from phase 1 to 4
> Sending ident
> >>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
> DIS:
> Preferred octets: 256
> Can receive fax
> Supported data signalling rates: V.27ter and V.29
> R8x7.7lines/mm and/or 200x200pels/25.4mm OK
> 2D coding OK
> Scan line length: 215mm
> Recording length: A4 (297mm)
> Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
> R8x15.4lines/mm OK
> Minimum scan line time for higher resolutions: T15.4 = T7.7
> >>> DIS: 80 00 ce f0 80 80 01
> HDLC underflow in state 9
> Changed from phase 4 to 3
> Slow carrier up
> <<< TSI: 43 31 37 31 31 36 35 34 35 34 38 30 20 20 20 20 20 20 20 20 20
> TSI without final frame tag
> Remote fax gave TSI as: ""
> <<< DCS: 83 00 86 90 00
> DCS with final frame tag
> In state 9
> DCS:
> Can receive fax
> Selected data signalling rate: V.29, 9600bps
> 2D coding OK
> Scan line length: 215mm
> Recording length: A4 (297mm)
> Minimum scan line time: 5ms
> Get at 9600
> Changed from phase 3 to 5
> Fast carrier up
> Fast carrier down
> Fast carrier up
> Coarse carrier frequency 1699.90 (64)
> Training error 56.874846
> Training succeeded (constellation mismatch 44.212022)
> Fast carrier trained
> Fast carrier down
> Trainability test failed - longest run of zeros was 14
> >>> FTT: 44
> Fast carrier up
> Training failed (sequence failed)
> Fast carrier training failed
> Fast carrier down
> Fast carrier up
> Coarse carrier frequency 1700.33 (64)
> Training error 51.989152
> Training succeeded (constellation mismatch 37.988826)
> Fast carrier trained
> Fast carrier down
> Trainability test failed - longest run of zeros was 15
> >>> FTT: 44
> Fast carrier up
> Training failed (sequence failed)
> Fast carrier training failed
> Fast carrier down
> Fast carrier up
> Coarse carrier frequency 1700.32 (64)
> Training error 60.898646
> Training succeeded (constellation mismatch 46.138793)
> Fast carrier trained
> Fast carrier down
> Trainability test failed - longest run of zeros was 17
> >>> FTT: 44
> Fast carrier up
> Training failed (sequence failed)
> Fast carrier training failed
> Fast carrier down
> Fast carrier up
> Coarse carrier frequency 1795.61 (4)
> Fast carrier down
> Fast carrier up
> Coarse carrier frequency 1789.60 (4)
> Fast carrier down
> -- Channel 0/1, span 1 got hangup
> -- Hungup 'Zap/1-1'
> 
> 
> 
> 
> 
> 

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RE: [Asterisk-Users] E1 config help and guidance

[Steve Hanselman]

Doesn't matter which span you use, but the Telco span should be set as clock
1 and CPE, the Meridian span should be set as clock 0 and NET.

Use only the channels you have been assigned on the Telco end, you as many
channels as you want on the Meridian end.

Take a look to see the order that inbound calls come from the Telco and
outbound calls come from the Meridian (just place 2 calls sequentially with
pri debug span x), then make sure you allocate the channels accordingly
(there's an option in the Group for zaptel for top down etc down with the
character that precedes the group number).

That's about it.

Steve


-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED] 
Sent: 09 July 2004 19:00
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] E1 config help and guidance

I've googled / voip-info'd / searched until my eyes are blurry, but couldn't
see the info I was looking for. I've turned here for help!

Asterisk CVS head (9/7/04)
Fedora Core 2 (updated to 2.6.6 kernel)
DE405P (jumpers set to E1)

I want to put asterisk in the middle of our current pbx (Meridian Option11)

Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into a
euroISDN bearer. This bearer only has 10 channels activated (out of the 30).
Obviously, this works - handsets make external calls.

What I wanted to do was to add * to the mix, in the middle so that it can
intercept inbound / outbound calls and do what it needs to do, as well as
providing all the extra functionality that this wonderful product provides.

In order to achieve this, I assumed that I needed to take rj45 from the
bearer box and plug that into span 2, and take a cable from span 1 into the
bearer box.

My problem (and blurry eyes) come from not understanding the various
protocols to assign to each span. I want the meridian to think that it's
still plugged into the EuroISDN bearer. So span 2 should be set up as a
EuroISDN link ? What should span 1 be set up as ? What channels should be
configured ?

Any guidance (I'm not looking for the solution (would be nice!) but for
pointers in the right direction).

I have previously been able to set up asterisk using the x100p and graduated
to BRI isdn. I just got the 405 today and wanted to play!

Thanks in advance.

Julian.

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RE: [Asterisk-Users] Dell 6450 / TE405p

[Steve Hanselman]

Well, you should see an entry like this:
  Bus  0, device  11, function  0:
Communication controller: PCI device 10ee:0314 (Xilinx Corporation) (rev
1).
  IRQ 5.
  Master Capable.  Latency=64.  
  Non-prefetchable 32 bit memory at 0xda001000 [0xda00107f].

Any curious messages in dmesg when the machine is booted, any settings in
the bios related to PCI?

At least from this point you can discount any zaptel issues as this shows
regardless of whether zaptel is loaded or not.

Steve



-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED] 
Sent: 09 July 2004 15:30
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

I've attached the /proc/pci below, but I think it's hardware related, not os
- the dell does not seem to recognise that there is a card in the slot. Or
any slot I put it in :(

Thanks for the help, though.

Julian.

[EMAIL PROTECTED] root]# cat /proc/pci
PCI devices found:
  Bus  0, device   0, function  0:
Host bridge: ServerWorks CNB20HE Host Bridge (rev 33).
  Master Capable.  Latency=32.
  Bus  0, device   0, function  1:
Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 1).
  Master Capable.  Latency=32.
  Bus  0, device   0, function  2:
Host bridge: ServerWorks CNB20HE Host Bridge (rev 0).
  Master Capable.  Latency=32.
  Bus  0, device   0, function  3:
Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 0).
  Master Capable.  Latency=32.
  Bus  0, device   4, function  0:
VGA compatible controller: ATI Technologies Inc 3D Rage IIC (rev 122).
  Master Capable.  Latency=32.  Min Gnt=8.
  Prefetchable 32 bit memory at 0xfc00 [0xfcff].
  I/O at 0xec00 [0xecff].
  Non-prefetchable 32 bit memory at 0xfbeff000 [0xfbef].
  Bus  0, device   8, function  0:
Ethernet controller: Intel Corp. 82557/8/9 [Ethernet Pro 100] (rev 8).
  IRQ 26.
  Master Capable.  Latency=32.  Min Gnt=8.Max Lat=56.
  Non-prefetchable 32 bit memory at 0xfbefe000 [0xfbefefff].
  I/O at 0xe8c0 [0xe8ff].
  Non-prefetchable 32 bit memory at 0xfbd0 [0xfbdf].
  Bus  0, device  15, function  0:
ISA bridge: ServerWorks OSB4 South Bridge (rev 80).
  Bus  0, device  15, function  1:
IDE interface: ServerWorks OSB4 IDE Controller (rev 0).
  Master Capable.  Latency=64.
  I/O at 0x8b0 [0x8bf].
  Bus  3, device   9, function  0:
PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (rev 0).
  Master Capable.  Latency=32.  Min Gnt=6.
  Bus  4, device   0, function  0:
PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (#2) (rev 0).
  Master Capable.  Latency=32.  Min Gnt=6.
  Bus  4, device   1, function  0:
SCSI storage controller: QLogic Corp. ISP12160 Dual Channel Ultra3 SCSI
Proc
essor (rev 6).
  IRQ 27.
  Master Capable.  Latency=32.  Min Gnt=64.
  I/O at 0xcc00 [0xccff].
  Non-prefetchable 32 bit memory at 0xfaaff000 [0xfaaf].
  Bus  5, device   0, function  0:
RAID bus controller: American Megatrends Inc. MegaRAID (rev 32).
  IRQ 23.
  Master Capable.  Latency=32.
  Prefetchable 32 bit memory at 0xf000 [0xf7ff].
[EMAIL PROTECTED] root]#
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 09 July 2004 14:14
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

What have you got in /proc/pci?

Do you have to do anything funny on the Dell to tell it that a card is
there?  Maybe it has some kind of health monitoring that you can switch off?

-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED]
Sent: 09 July 2004 13:36
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

No, the modules don't load (no such device) and it doesn't show up on the
interrupts.

I think it's more basic than that - on the 6450 each slot has a led
indicating that there is a card present. Whatever slot I put the card in on
the 6450, it flashes on boot, and then goes out. If I put *any* of the other
PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then
the led goes on and stays on. It's as if the 6450 doesn't "like" the TE405P
and disables the slot.

On the 4400, the led goes on and stays on, and the TE405P flashes red on
each channel.

Julian.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 09 July 2004 13:16
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

Do the modules load (lsmod).

Are the interrupts assigned and unique (cat /proc/interrupts)

-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED]
Sent: 09 July 2004 13:17
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

Yeah, that's what I thought.

Some more developments - I've just plugged it into a Dell 4400 server,
worked first time. So what

RE: [Asterisk-Users] Dell 6450 / TE405p

[Steve Hanselman]

What have you got in /proc/pci?

Do you have to do anything funny on the Dell to tell it that a card is
there?  Maybe it has some kind of health monitoring that you can switch off?

-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED] 
Sent: 09 July 2004 13:36
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

No, the modules don't load (no such device) and it doesn't show up on the
interrupts.

I think it's more basic than that - on the 6450 each slot has a led
indicating that there is a card present. Whatever slot I put the card in on
the 6450, it flashes on boot, and then goes out. If I put *any* of the other
PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then
the led goes on and stays on. It's as if the 6450 doesn't "like" the TE405P
and disables the slot.

On the 4400, the led goes on and stays on, and the TE405P flashes red on
each channel.

Julian.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 09 July 2004 13:16
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

Do the modules load (lsmod).

Are the interrupts assigned and unique (cat /proc/interrupts)

-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED]
Sent: 09 July 2004 13:17
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

Yeah, that's what I thought.

Some more developments - I've just plugged it into a Dell 4400 server,
worked first time. So what's with the 6450 ? 

I've got another spare 6450 kicking around somewhere - I'm going to try that
one as well.

Thanks.

Julian. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 09 July 2004 13:01
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

If the card plugs into the slot then it's not a power (3.3v/5v) issue as the
cards are physically different.


-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED]
Sent: 09 July 2004 13:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dell 6450 / TE405p

I'm having some trouble here - need some help! 

I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell
6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses
and one 32-bit bus Expansion slots  seven hot-pluggable PCI slots (two
64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)"

I cannot get the card working in any of the slots. When I power the server
up, the digium card blinks red for a second, and the dell pci light goes
out. However, if I plug it into a standard dell dimension, I get the 4
channels flashing red when I power the machine up.

My understanding of the pci bus is (obviously) limited - why doesn't the
card work in the server ? Should I have bought the te410p instead ? 

Help!

Julian.

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RE: [Asterisk-Users] Dell 6450 / TE405p

[Steve Hanselman]

Do the modules load (lsmod).

Are the interrupts assigned and unique (cat /proc/interrupts)

-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED] 
Sent: 09 July 2004 13:17
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

Yeah, that's what I thought.

Some more developments - I've just plugged it into a Dell 4400 server,
worked first time. So what's with the 6450 ? 

I've got another spare 6450 kicking around somewhere - I'm going to try that
one as well.

Thanks.

Julian. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 09 July 2004 13:01
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p

If the card plugs into the slot then it's not a power (3.3v/5v) issue as the
cards are physically different.


-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED]
Sent: 09 July 2004 13:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dell 6450 / TE405p

I'm having some trouble here - need some help! 

I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell
6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses
and one 32-bit bus Expansion slots  seven hot-pluggable PCI slots (two
64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)"

I cannot get the card working in any of the slots. When I power the server
up, the digium card blinks red for a second, and the dell pci light goes
out. However, if I plug it into a standard dell dimension, I get the 4
channels flashing red when I power the machine up.

My understanding of the pci bus is (obviously) limited - why doesn't the
card work in the server ? Should I have bought the te410p instead ? 

Help!

Julian.

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RE: [Asterisk-Users] Dell 6450 / TE405p

[Steve Hanselman]

If the card plugs into the slot then it's not a power (3.3v/5v) issue as the
cards are physically different.


-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED] 
Sent: 09 July 2004 13:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dell 6450 / TE405p

I'm having some trouble here - need some help! 

I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell
6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses
and one 32-bit bus Expansion slots  seven hot-pluggable PCI slots (two
64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)"

I cannot get the card working in any of the slots. When I power the server
up, the digium card blinks red for a second, and the dell pci light goes
out. However, if I plug it into a standard dell dimension, I get the 4
channels flashing red when I power the machine up.

My understanding of the pci bus is (obviously) limited - why doesn't the
card work in the server ? Should I have bought the te410p instead ? 

Help!

Julian.

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[Asterisk-Users] Sending SABME continuosly. Urgent help needed!

[Steve Hanselman]

Hi David, Sorry for the very late reply to this, but it looks as though you're in PRI_NET rather than PRI_CPE, I believe that the network sends to a SABME and you reply with an unnumbered response, Q921 is then established?  Regards Steve   David Morillo wrote: Hi, I'm trying to install an E1 PRI, and I need it working byMonday, but although everything seems ok, I get no response to calls. When I make a pri extense debug on span 1, I repeatedly get thefollowing:    Sending Set Asynchronous Balanced Mode Extended    [ 00 01 7f ]    Unnumbered frame:    SAPI: 00 C/R: 0 EA: 0 TEI: 999      EA: 1  M3: 3  P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balancedmode extended]    0 bytes of data.    And nothing else.    When making a call to that E1, I see the message "D-Channel onspan 1 up" 4 times, and then a Informational frame, with TEI:000 EA:1 and anything else with zero (13 bytes of data).    Stopping T_203 timer    Starting T_200 timer    Protocol Discriminator: Q.931 (8) len=13    Call Ref: len= 2 (reference 0/0x0) (Originator)    Message type: RESTART (70)    Channel ID (len= 5) [Ext: 1  IntID: Implicit, PRI Spare: 0,Exclusive Dchan:0    Chan Sel Reserved    Ext: 1  Coding:0  Number Specified   Channel Type: 3    Ext: 1  Spare: o  Resetting Inidicated Channel (0) ]    Then "D-Channel on span 1 downn", and finally, after a while:    (...) Warning[11276]: chan_zap.c:5993 zt_pri_error: PRI: Read on46 failed: Unknown error 500    (...) Notice[11276]: chan_zap.c:6708 pri_dchannel: PRI gotevent: 8 on span 1    I think I have Asterisk stable version 1.0, CVS updated today    Can anyone help me? Please! :S Zaptel.conf    --    span=1,1,0,ccs,hdb3    bchan=1-15    dchan=16    bchan=17-31 loadzone = es    defaultzone=es    Zapata.conf:    -- [channels]    language=es    context=default    switchtype=euroisdn    signalling=pri_cpe    usecallerid=yes    hidecallerid=no    callwaiting=yes    usecallingpres=yes    callwaitingcallerid=yes    threewaycalling=yes    transfer=yes    cancallforward=yes    callreturn=yes    echocancel=yes    echocancelwhenbridged=yes    rxgain=0.0    txgain=0.0    callgroup=1    pickupgroup=1    immediate=no    jitterbuffers=4    group = 1    channel => 1-15,17-31 I have also tried with    span=1,0,0,ccs,hdb3    immediate = yes    The line has not CRC activated (I have asked)    Thanks!

 

 

Steve Hanselman

Brendata (UK) Ltd

 

Tel: +44 (0)1268 466111

Fax: +44 (0)870 1387283

Mob: +44 (0)7973 750993

 

 

 






The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk

RE: [Asterisk-Users] Penalty in queues.conf

[Steve Hanselman]

It's so you can have agents that are less likely to take calls (e.g. imagine
a sales queue, you'd have the sales people with no penalty, you might have
the receptionists with a penalty of 1 and us propeller heads in technical
support with a penalty of 2).

The technical support people would only be offered a call from the sales
queue if all the sales people and the receptionists were busy.

Steve


-Original Message-
From: Isamar Maia [mailto:[EMAIL PROTECTED] 
Sent: 04 July 2004 10:52
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Penalty in queues.conf



I have already read explanation about that in some places but I don't have
still a clear image about the meaning of Penalty parameter inside of
queues.conf
What means that?

Thanks,

Isamar


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RE: [Asterisk-Users] Cisco 7960G and *

[Steve Hanselman]

It'll work, either as a SIP phone with the SIP image, or as skinny using
wither chan_sccp or chan_skinny (check the wiki).

Steve


-Original Message-
From: Matt Davies | MattDavies.Net [mailto:[EMAIL PROTECTED] 
Sent: 02 July 2004 15:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7960G and *

I have been doing so much reading on phones lately that I have completely
lost track of some things. I seem to remember that there was one series of
Cisco IP phones that required Cisco's call manager. Does anyone know if the
7960 will work with Asterisk or does it require call manager?

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RE: [Asterisk-Users] Providing Telewest in the UK with per extens ion outbound callerID

[Steve Hanselman]

When the original PBX was installed we
asked them to override the CLI and provide a single number as the PBX couldn't
provide the DDI number, now the contact at Telewest believes it's
somewhere between illegal and impossible to provide DDI numbers to the outside
world.

 

 

-Original
Message-
From: Storer, Darren
[mailto:[EMAIL PROTECTED] 
Sent: 01 July 2004 10:13
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Providing Telewest in the UK with per extension outbound callerID

 



Hi
Steve,





 





Telewest
should already allow the CLI transmission of your DDI range, without further
datafill changes. If it doesn't work you should check that you are sending the
appropriate number of digits.





 





Try
sending:





 





 - 3
digit CLI





 - the
whole number (minus the leading zero)





 





If the
comments above don't help please post a trace of an outgoing call and detail
the number, if any, that is presented to the Called Party.





 





HTH





 





Darren





-- 





Comgate





Telco>Internet





 





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of Steve Hanselman
Sent: 01 July 2004 09:57
To:
'[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users]
Providing Telewest in the UK with per extension outbound callerID

Would be
nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just
our DDI if that was all I could get.

 

Steve

 

 

-Original
Message-
From: Storer, Darren
[mailto:[EMAIL PROTECTED] 
Sent: 01 July 2004 09:35
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Providing Telewest in the UK with per extension outbound callerID

 



Hi
Steve,





 





SH>
Is anybody in the UK using Telewest as a PRI Telco provider?



SH>
Are you sending them caller ID?

 

Just a
quick point of clarification before commenting further, do you wish to make
calls via Telewest's network and send the CLI of your own DDI number range or
do you wish to send "other numbers" as your CLI? If you are seeking to achieve
the latter, what sort of numbers do you wish to propagate as the CLI for
your calls?

 

Regards


Darren

-- 

Comgate

Telco>Internet



-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve Hanselman
Sent: 30 June 2004 18:57
To:
'[EMAIL PROTECTED]'
Subject: [Asterisk-Users]
Providing Telewest in the UK with per extension outbound callerID

Hi,

 

Is anybody in the UK using Telewest
as a PRI Telco provider?

 

Are you sending them caller ID?

 

I've been told by Telewest that:-

 

1.  Oftel
doesn't allow them to accept caller ID (this is rubbish, and I replied pointing
out where in the link to Oftel that they sent me it was stated.  We need
Type 2 caller ID) 

2.  Telewest
can't do this. (this is rubbish, I'm certain that some of our customers use
Telewest and they provide them with caller ID) 

 

So, does anybody do this, and if so,
what did you have to request from them in order to enable it, and what do you
provide to them (how many digits and in what format).

 

Regards

 

Steve



The information
contained in this email is intended for the personal and confidential use
of the addressee only. It may also be privileged information. If you are not
the intended
recipient then you are hereby notified that you have received this document in
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that any review, distribution or copying of this document is strictly
prohibited. If you have 
received this communication in error, please notify Brendata immediately on: 

+44 (0)1268 466100, or email '[EMAIL PROTECTED]' 

Brendata (UK) Ltd
Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK
Registered Office as above. Registered in England No. 2764339

See our current vacancies at www.brendata.co.uk








The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk

RE: [Asterisk-Users] Providing Telewest in the UK with per extens ion outbound callerID

[Steve Hanselman]

Would be nice to do both (type 2 and 3 I
believe in Oftel terms), but I'd accept just our DDI if that was all I
could get.

 

Steve

 

 

-Original
Message-
From: Storer, Darren
[mailto:[EMAIL PROTECTED] 
Sent: 01 July 2004 09:35
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Providing Telewest in the UK with per extension outbound callerID

 



Hi
Steve,





 





SH>
Is anybody in the UK using Telewest as a PRI Telco provider?



SH>
Are you sending them caller ID?

 

Just a
quick point of clarification before commenting further, do you wish to make
calls via Telewest's network and send the CLI of your own DDI number range or
do you wish to send "other numbers" as your CLI? If you are seeking
to achieve the latter, what sort of numbers do you wish to propagate
as the CLI for your calls?

 

Regards


Darren

-- 

Comgate

Telco>Internet



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of Steve Hanselman
Sent: 30 June 2004 18:57
To:
'[EMAIL PROTECTED]'
Subject: [Asterisk-Users]
Providing Telewest in the UK with per extension outbound callerID

Hi,

 

Is anybody in the UK using Telewest
as a PRI Telco provider?

 

Are you sending them caller ID?

 

I've been told by Telewest that:-

 

1. 
Oftel doesn't allow them to accept caller ID (this is
rubbish, and I replied pointing out where in the link to Oftel that they sent
me it was stated.  We need Type 2 caller ID) 

2. 
Telewest can't do this. (this is rubbish, I'm certain
that some of our customers use Telewest and they provide them with caller ID)


 

So, does anybody do this, and if so,
what did you have to request from them in order to enable it, and what do you
provide to them (how many digits and in what format).

 

Regards

 

Steve








The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk

[Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID

[Steve Hanselman]

Hi,

 

Is anybody in the UK using Telewest as
a PRI Telco provider?

 

Are you sending them caller ID?

 

I've been told by Telewest that:-

 


 Oftel doesn't allow them to accept caller ID
 (this is rubbish, and I replied pointing out where in the link to Oftel
 that they sent me it was stated.  We need Type 2 caller ID)
 Telewest can't do this. (this is rubbish, I'm
 certain that some of our customers use Telewest and they provide them with
 caller ID)


 

So, does anybody do this, and if so, what did you have to
request from them in order to enable it, and what do you provide to them (how
many digits and in what format).

 

Regards

 

Steve

 

 

 






The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk

RE: [Asterisk-Users] How to test E1 interfacing?

[Steve Hanselman]

1. Can't answer, sorry!
2. Yes, although you have to be clear about what you're proving, you're
proving that the cards work, not that asterisk is correctly configured and
will eventually talk exactly correctly to the entity that you'll connect it
to be a PBX or a telco.  You'd configure one as net and the other as cpe,
and you could then back to back them.  We did this.

Steve (based in the UK)


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: 29 June 2004 10:00
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] How to test E1 interfacing?

Hi,

I have a project coming up which will need to interface Asterisk to
E1 trunks in the UK. I have a couple of questions which I hope someone
can answer, or give me some pointers:

1. If I want two E1 trunks, is there anything to choose, performance-wise,
   between using two ports on a single TE405P, and using two E100P cards?

2. How can I test the E1 operation in the lab, which doesn't have an
   E1 line available, before taking the unit to the installation site?
   Can I run two Asterisks back-to-back? Can I run one port into another
   on a single TE405P?

I couldn't find anything on the above in the Wiki; if I didn't look hard
enough, please tell me where I missed. Thanks.

Any advice would be gratefully received!

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Queue hold time in seconds

[Steve Hanselman]

I'm going to modify the queue announcements to allow
for rounded seconds (e.g. we want to know to the tens of seconds.  E.g.
Average wait 1 minute 20 seconds).

 

I'm going to add the optional announce of seconds to
the queue config and a rounding factor (e.g. 10 in our case).

 

The following parameters will be added

 

Queue-announce-seconds (default is off)

Queue-seconds (default will be an as yet unrecorded "queue-seconds")

Queue-rounding-seconds (default will be 10)

 

Have I missed anything?

 

Steve

 






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