Re: [sipx-users] Generate CSR Question

[Tony Graziano]

It is on the tracker. I think the GUI needs to be adjusted to allow it. I
also know 4.6 has major cert rework, maybe it can be posted to the dev list
whether it is being addressed in 4.6 or not.

http://track.sipfoundry.org/browse/XX-9390

All of the above (intermediate and changing the script to 2048 bits is on
the mailing list. It should be on the wiki AND in the JIRA though.

It's not like its someones secret sauce or anything.

On Fri, Apr 20, 2012 at 5:09 PM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:

> I also had to drop the Certificate Authorities CRT files for GoDaddy of
> gd-class2-root.crt, gd_intermediate.crt & gdroot-g2.crt into the(
>  /etc/sipxpbx/ssl/authorities ) directory. I restarted the sipxecs service
> and then proceeded to add the web certificate downloaded from GoDaddy.
>
> ** **
>
> sipXecs System/Certificate Authorities area would not allow me to add the
> CA CRT files for GoDaddy via the web administration portal. That is why I
> published the above information.
>
> ** **
>
> Thanks everyone…
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Robert Schroeder
> *Sent:* Friday, April 20, 2012 4:50 PM
> *To:* sipx-users@list.sipfoundry.org
>
> *Subject:* Re: [sipx-users] Generate CSR Question
>
> ** **
>
> Yeps, no luck in the search.
>
> ** **
>
> However Jim Nolen of IIPS was a great help and gave me the following
> information to solve the problem.
>
> ** **
>
> Edit: /usr/bin/ssl-cert/gen-ssl-keys.sh:
>
> ServerKeyBits=1024[change to 2048]
>
> ** **
>
> If I knew how to add this info to the wiki I would. Perhaps a feature
> could be added to ask the user hitting the generate button if they would
> like a 1024, 2048 or 4096 CSR.
>
> ** **
>
> Thanks Mr. Nolen for the help (Smiles)
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Michael Picher
> *Sent:* Friday, April 20, 2012 4:35 PM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] Generate CSR Question
>
> ** **
>
> did you check the wiki?
>
> On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder <
> robert.schroe...@memberfirstmortgage.com> wrote:
>
> How do I change the configuration for the certificates area to generate a
> 2048 bit key instead of a 1024? I have changed the openssl.cnf file in
> /etc/pki/tls/ location and selected the generate button and still no 2048
> key is generated.
>
>  
>
> I am sure this is an educational issue on my part.
>
>  
>
> Yes I have searched the wiki site.
>
>  
>
> Thanks everyone,
>
>  
>
> Rob
>
> ** **
> --
>
>
> NOTICE: This electronic mail message and any content within it are intended 
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> entities to which it is addressed. The message, together with any attachments 
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> print, save, copy, disclosure or distribution
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> is strictly prohibited. If you have received this message in error, please 
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>
>
>
> 
>
> ** **
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> --
> Michael Picher, Director of Technical Services
> eZuce, Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
>
> O.978-296-1005 X2015
> M.207-956-0262
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>
> ** **
>
>
> 
> 
>
> There are 10 kinds of people in the world, those who understand binary and
> those who don't.
>
> ** **
>
> ** **
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> NOTICE: This electronic mail message and any content within it are intended 
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> entities to which it is addressed. The message, together with any attachments 
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Re: [sipx-users] Generate CSR Question

[Robert Schroeder]

I also had to drop the Certificate Authorities CRT files for GoDaddy of 
gd-class2-root.crt, gd_intermediate.crt & gdroot-g2.crt into the(  
/etc/sipxpbx/ssl/authorities ) directory. I restarted the sipxecs service and 
then proceeded to add the web certificate downloaded from GoDaddy.
 
sipXecs System/Certificate Authorities area would not allow me to add the CA 
CRT files for GoDaddy via the web administration portal. That is why I 
published the above information.
 
Thanks everyone...
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert Schroeder
Sent: Friday, April 20, 2012 4:50 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Generate CSR Question
 
Yeps, no luck in the search.
 
However Jim Nolen of IIPS was a great help and gave me the following 
information to solve the problem.
 
Edit: /usr/bin/ssl-cert/gen-ssl-keys.sh:
ServerKeyBits=1024[change to 2048]
 
If I knew how to add this info to the wiki I would. Perhaps a feature could be 
added to ask the user hitting the generate button if they would like a 1024, 
2048 or 4096 CSR.
 
Thanks Mr. Nolen for the help (Smiles)
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Friday, April 20, 2012 4:35 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Generate CSR Question
 
did you check the wiki?
On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder 
 wrote:
How do I change the configuration for the certificates area to generate a 2048 
bit key instead of a 1024? I have changed the openssl.cnf file in /etc/pki/tls/ 
location and selected the generate button and still no 2048 key is generated.
 
I am sure this is an educational issue on my part.
 
Yes I have searched the wiki site.
 
Thanks everyone,
 
Rob
 


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-- 
Michael Picher, Director of Technical Services
eZuce, Inc.
300 Brickstone Square
Suite 201
Andover, MA. 01810
O.978-296-1005 X2015 
M.207-956-0262
@mpicher  
www.ezuce.com
 

There are 10 kinds of people in the world, those who understand binary and 
those who don't.
 
 


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Re: [sipx-users] Generate CSR Question

[jnolen]

Rob,

I'd appreciate it if you would not publish my name.

Thanks,

jim


> Yeps, no luck in the search.
> 
>  
> 
> However Jim Nolen of IIPS was a great help and gave me the following
> information to solve the problem.
> 
>  
> 
> Edit: /usr/bin/ssl-cert/gen-ssl-keys.sh:
> 
> ServerKeyBits=1024[change to 2048]
> 
>  
> 
> If I knew how to add this info to the wiki I would. Perhaps a feature
> could be added to ask the user hitting the generate button if they
> would like a 1024, 2048 or 4096 CSR.
> 
>  
> 
> Thanks Mr. Nolen for the help (Smiles)
> 
>  
> 
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
> Picher
> Sent: Friday, April 20, 2012 4:35 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Generate CSR Question
> 
>  
> 
> did you check the wiki?
> 
> On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder
>  wrote:
> 
> How do I change the configuration for the certificates area to
> generate a 2048 bit key instead of a 1024? I have changed the
> openssl.cnf file in /etc/pki/tls/ location and selected the generate
> button and still no 2048 key is generated.
> 
>  
> 
> I am sure this is an educational issue on my part.
> 
>  
> 
> Yes I have searched the wiki site.
> 
>  
> 
> Thanks everyone,
> 
>  
> 
> Rob
> 
> 
>  
> 
>
> __
> NOTICE: This electronic mail message and any content within it are intended 
> exclusively for the individual(s) or 
> entities to which it is addressed. The message, together with any attachments 
> and all other content, may contain
> confidential and/or privileged information. Any unauthorized review, use, 
> print, save, copy, disclosure or distribution
> is strictly prohibited. If you have received this message in error, please 
> immediately advise the sender by reply email
> and delete all copies.
> 
> 
> 
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
> 
> 
> 
> 
>  
> 
> 
> -- 
> Michael Picher, Director of Technical Services
> eZuce, Inc.
> 
> 300 Brickstone Square
> 
> Suite 201
> 
> Andover, MA. 01810
> 
> O.978-296-1005 X2015 
> M.207-956-0262
> @mpicher  
> www.ezuce.com
> 
> 
>  
> 
> 
> 
> 
> 
> There are 10 kinds of people in the world, those who understand binary
> and those who don't.
> 
> 
>  
> 
> 
> 
> 
> 
> __
> NOTICE: This electronic mail message and any content within it are intended 
> exclusively for the individual(s) or 
> entities to which it is addressed. The message, together with any attachments 
> and all other content, may contain
> confidential and/or privileged information. Any unauthorized review, use, 
> print, save, copy, disclosure or distribution
> is strictly prohibited. If you have received this message in error, please 
> immediately advise the sender by reply email
> and delete all copies.
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> List Archive: http://list.sipfoundry.org/archive/sipx-users/


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Re: [sipx-users] Generate CSR Question

[Robert Schroeder]

Yeps, no luck in the search.
 
However Jim Nolen of IIPS was a great help and gave me the following 
information to solve the problem.
 
Edit: /usr/bin/ssl-cert/gen-ssl-keys.sh:
ServerKeyBits=1024[change to 2048]
 
If I knew how to add this info to the wiki I would. Perhaps a feature could be 
added to ask the user hitting the generate button if they would like a 1024, 
2048 or 4096 CSR.
 
Thanks Mr. Nolen for the help (Smiles)
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Friday, April 20, 2012 4:35 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Generate CSR Question
 
did you check the wiki?
On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder 
 wrote:
How do I change the configuration for the certificates area to generate a 2048 
bit key instead of a 1024? I have changed the openssl.cnf file in /etc/pki/tls/ 
location and selected the generate button and still no 2048 key is generated.
 
I am sure this is an educational issue on my part.
 
Yes I have searched the wiki site.
 
Thanks everyone,
 
Rob
 


NOTICE: This electronic mail message and any content within it are intended 
exclusively for the individual(s) or 
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confidential and/or privileged information. Any unauthorized review, use, 
print, save, copy, disclosure or distribution
is strictly prohibited. If you have received this message in error, please 
immediately advise the sender by reply email
and delete all copies.

___
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-- 
Michael Picher, Director of Technical Services
eZuce, Inc.
300 Brickstone Square
Suite 201
Andover, MA. 01810
O.978-296-1005 X2015 
M.207-956-0262
@mpicher  
www.ezuce.com
 

There are 10 kinds of people in the world, those who understand binary and 
those who don't.
 


NOTICE: This electronic mail message and any content within it are intended 
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Re: [sipx-users] Generate CSR Question

[Tony Graziano]

I seem to recall the script may need to be or was it already modified to
handle 2048 bit certificates?

Besides that I think it had to be done manually AND noone updated the wiki
or the list as to whether or not it worked.

On Fri, Apr 20, 2012 at 4:34 PM, Michael Picher  wrote:

> did you check the wiki?
>
> On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder <
> robert.schroe...@memberfirstmortgage.com> wrote:
>
>> How do I change the configuration for the certificates area to generate a
>> 2048 bit key instead of a 1024? I have changed the openssl.cnf file in
>> /etc/pki/tls/ location and selected the generate button and still no 2048
>> key is generated.
>>
>> ** **
>>
>> I am sure this is an educational issue on my part.
>>
>> ** **
>>
>> Yes I have searched the wiki site.
>>
>> ** **
>>
>> Thanks everyone,
>>
>> ** **
>>
>> Rob
>>
>>
>> --
>>
>> NOTICE: This electronic mail message and any content within it are intended 
>> exclusively for the individual(s) or
>>
>> entities to which it is addressed. The message, together with any 
>> attachments and all other content, may contain
>>
>> confidential and/or privileged information. Any unauthorized review, use, 
>> print, save, copy, disclosure or distribution
>>
>> is strictly prohibited. If you have received this message in error, please 
>> immediately advise the sender by reply email
>> and delete all copies.
>>
>> ___
>> sipx-users mailing list
>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
> --
> Michael Picher, Director of Technical Services
> eZuce, Inc.
>
> 300 Brickstone Square
>
> Suite 201
>
> Andover, MA. 01810
> O.978-296-1005 X2015
> M.207-956-0262
> @mpicher 
> www.ezuce.com
>
>
> 
> There are 10 kinds of people in the world, those who understand binary and
> those who don't.
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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~~
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Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
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Re: [sipx-users] Generate CSR Question

[Michael Picher]

did you check the wiki?

On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:

> How do I change the configuration for the certificates area to generate a
> 2048 bit key instead of a 1024? I have changed the openssl.cnf file in
> /etc/pki/tls/ location and selected the generate button and still no 2048
> key is generated.
>
> ** **
>
> I am sure this is an educational issue on my part.
>
> ** **
>
> Yes I have searched the wiki site.
>
> ** **
>
> Thanks everyone,
>
> ** **
>
> Rob
>
>
> --
>
> NOTICE: This electronic mail message and any content within it are intended 
> exclusively for the individual(s) or
>
> entities to which it is addressed. The message, together with any attachments 
> and all other content, may contain
>
> confidential and/or privileged information. Any unauthorized review, use, 
> print, save, copy, disclosure or distribution
>
> is strictly prohibited. If you have received this message in error, please 
> immediately advise the sender by reply email
> and delete all copies.
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



-- 
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eZuce, Inc.

300 Brickstone Square

Suite 201

Andover, MA. 01810
O.978-296-1005 X2015
M.207-956-0262
@mpicher 
www.ezuce.com


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Re: [sipx-users] Fax Problems (Big Time)

[Robert Schroeder]

So far I have had no reports of problems. I am waiting for a bit to know if 
this issue is resolved completely before posting details of the AC 
configuration.
 
So Far So Good,
 
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 12:10 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
 
did you do that? if so, did it fix it?
On Fri, Apr 20, 2012 at 11:53 AM, Robert Schroeder 
 wrote:
Tony,
 
Thank you for the suggestion and for your help.
 
Rob
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 11:00 AM

To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
 
Sorry, am not a audiocodes guru. I have documented the Patton PRi gateways for 
use with t.38 in the wiki. Perhaps someone else would be so kind as to do the 
same for an AC PRI gateway instead of keeping it a big secret and make sure it 
is also documented on the wiki.
 
Try:
 
Configuration > Protocol Configuration > Protocol Definition > SIP General 
Parameters
Fax Signaling Method T.38 Relay
Detect Fax on Answer Tone Initiate T.38 on Preamble
SIP Transport Type UDP
 
I don't know what the other settings are for Detect Fax on Answer Tone, you 
really should address this before logging and tracing, because it should have 
fax set for t.38 on the Mediant FIRST.
 
For sanity, when you do find Nirvana, please post it here for someone to update 
the wiki if you don't have a wiki account.
 
On Fri, Apr 20, 2012 at 9:50 AM, Robert Schroeder 
 wrote:
Tony:
 
I did find a Fax Signaling Method of T.38 Relay under the Coders And Profile 
Definitions - IP Profile Settings - Gateway Parameters - Fax Signaling Method.
 
 
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 9:29 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
 
faxing into sipx only support t.38 and your audiocodes fax settings have g711u. 
you need to change it to t.38.
On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder 
 wrote:
Has anyone had problems with faxing into sipXecs? We are having huge problems 
with faxes being delivered to the users emails as zero page tiff's. It is not 
an every fax occurrence however it is at a rate that I am no longer able to 
rely upon the faxing delivery capabilities of sipXecs. I am trying to determine 
if the problems is with the AudioCodes, PRI Provider or sipXecs.
 
Just wondering if there is a known issue. I am running (4.4.0- 
2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with 
an AudioCodes Mediant 1000 SIP Gateway.
 
Listed below is my current configuration settings on the AudioCodes!
 
AudioCodes Information
Version ID: 6.00A.032.003
DSP Type: 2
DSP Software Version: 6008
DSP Software Name: 624AE3
Flash Version: 208
 
AudioCodes Configuration - Media Settings - Fax/Modem/CID Settings
 
General Settings
Fax Transport Mode: RelayEnable
Caller ID Transport Type: Mute
Caller ID Type: Standard Bellcore
V.21 Modem Transport Type: Disable
V.22 Modem Transport Type: Disable
V.23 Modem Transport Type: Disable
V.32 Modem Transport Type: Disable
V.34 Modem Transport Type: Enable Bypass
Fax CNG Mode: Disable
CNG Detector Mode: Disable
 
Fax Relay Settings
Fax Relay Redundancy Depth: 0
Fax Relay Enhanced Redundancy Depth: 4
Fax Relay ECM Enable: Enable
Fax Relay Max Rate (bps): 9600bps
 
Bypass Settings
Fax/Modem Bypass Coder Type: G711Mulaw
Fax/Modem Bypass Packing Factor: 1
Fax Bypass Output Gain: 0
Modem Bypass Output Gain: 0
 
Any suggestions from the techs would be very welcomed!
 
Thanks Everyone,
 
Rob Schroeder
 


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[sipx-users] Generate CSR Question

[Robert Schroeder]

How do I change the configuration for the certificates area to generate a 2048 
bit key instead of a 1024? I have changed the openssl.cnf file in /etc/pki/tls/ 
location and selected the generate button and still no 2048 key is generated.
 
I am sure this is an educational issue on my part.
 
Yes I have searched the wiki site.
 
Thanks everyone,
 
Rob


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immediately advise the sender by reply email
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Re: [sipx-users] SharedAppearanceAgent does not match the software version (4.4.0).

[Kyle Haefner]

Dave,

Thanks!  I turned it off, not sure how it got started.  I updated the
wiki page to reflect this:

http://wiki.sipfoundry.org/display/sipXecs/Roles%2C+Services+and+Processes

Kyle

On Fri, Apr 20, 2012 at 12:24 PM, Dave Deutschman
 wrote:
> Kyle,
>
> The SharedAppearance Agent is not redundant and should only be running on the 
> master.  It will run on the secondary and Polycom phones will send 
> subscriptions to it and the process on the master.  However, it confuses the 
> phones and they stop handling shared line calls properly.
>
> There was a bug in the startup scripts of 4.2.0 where it started the process 
> on a secondary but that had been fixed in a 4.2.1 patch release by Nortel.  I 
> am not sure if that fix made it back to the open source builds.
>
> Sincerely,
>
> Dave Deutschman
> Managing Partner
> Innovational IP Solutions, LLC
>
> PO Box 983
> Bothell, WA  98041
>
> 206.965.9586 x 301 (o)
> 425.478.9642 (m)
> 3...@sipx.isdomain.net (s)
>
> www.innovational.net
>
> "The information transmitted is intended only for the person or entity to 
> which it is addressed and may contain proprietary, confidential and/or 
> legally privileged material. Any review, retransmission, dissemination or 
> other use of, or taking of any action in reliance upon, this information by 
> persons or entities other than the intended recipient is prohibited. If you 
> received this in error, please contact the sender and delete the material 
> from all computers."
>
>
> From: sipx-users-boun...@list.sipfoundry.org 
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner
> Sent: Friday, April 20, 2012 10:50 AM
> To: sipx-users
> Subject: [sipx-users] SharedAppearanceAgent does not match the software 
> version (4.4.0).
>
>
> Hi Users,
>
> I'm running Sipx 4.4 in a three node cluster with the most recent
> updates for 4.4.0- 2012-02-08EST09:10:08 ip-10-72-10-163.
>
>  In one of the redundant nodes when I get this:
>
> sudo sipxproc | grep SharedAppearanceAgent
>  "SharedAppearanceAgent"=>"Disabled",
>
> In the other redundant node i get this:
> sudo sipxproc | grep SharedAppearanceAgent
>  "SharedAppearanceAgent"=>"ConfigurationMismatch",
>
> sudo sipxproc -m SharedAppearanceAgent
> ["version.mismatch: software '4.4.0' != config ''"]
>
> So the question is, "Is SharedAppearanceAgent supposed to be running
> on redundant nodes"
>
> According to this it is supposed to be running.
> http://wiki.sipfoundry.org/display/sipXecs/Roles,+Services+and+Processes
>
>
> The next question is if it is supposed to be running, why does it come
> up with this error?  I've sent profiles many times, restarted sipx and
> even rebooted the server.
>
> Any ideas?
>
> Thanks!
>
> Kyle
>
>
> --
> Kyle Haefner, M.S.
> Communication Systems Programmer
> Colorado State University
> Fort Collins, CO
> Phone: [X] 970-491-1012
> Email:  kyle.haef...@colostate.edu
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
> -
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.1913 / Virus Database: 2411/4948 - Release Date: 04/20/12



-- 
Kyle Haefner, M.S.
Communication Systems Programmer
Colorado State University
Fort Collins, CO
Phone: 970-491-1012
Email:  kyle.haef...@colostate.edu
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Re: [sipx-users] SharedAppearanceAgent does not match the software version (4.4.0).

[Dave Deutschman]

Kyle,

 

The SharedAppearance Agent is not redundant and should only be running on
the master.  It will run on the secondary and Polycom phones will send
subscriptions to it and the process on the master.  However, it confuses the
phones and they stop handling shared line calls properly.

 

There was a bug in the startup scripts of 4.2.0 where it started the process
on a secondary but that had been fixed in a 4.2.1 patch release by Nortel.
I am not sure if that fix made it back to the open source builds.

 

Sincerely,

 

Dave Deutschman

Managing Partner

Innovational IP Solutions, LLC

 

PO Box 983

Bothell, WA  98041

 

206.965.9586  x 301 (o)

425.478.9642 (m)

  3...@sipx.isdomain.net (s)

 

www.innovational.net  

 

"The information transmitted is intended only for the person or entity to
which it is addressed and may contain proprietary, confidential and/or
legally privileged material. Any review, retransmission, dissemination or
other use of, or taking of any action in reliance upon, this information by
persons or entities other than the intended recipient is prohibited. If you
received this in error, please contact the sender and delete the material
from all computers." 

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner
Sent: Friday, April 20, 2012 10:50 AM
To: sipx-users
Subject: [sipx-users] SharedAppearanceAgent does not match the software
version (4.4.0).

 

Hi Users,

I'm running Sipx 4.4 in a three node cluster with the most recent
updates for 4.4.0- 2012-02-08EST09:10:08 ip-10-72-10-163.

 In one of the redundant nodes when I get this:

sudo sipxproc | grep SharedAppearanceAgent
 "SharedAppearanceAgent"=>"Disabled",

In the other redundant node i get this:
sudo sipxproc | grep SharedAppearanceAgent
 "SharedAppearanceAgent"=>"ConfigurationMismatch",

sudo sipxproc -m SharedAppearanceAgent
["version.mismatch: software '4.4.0' != config ''"]

So the question is, "Is SharedAppearanceAgent supposed to be running
on redundant nodes"

According to this it is supposed to be running.
http://wiki.sipfoundry.org/display/sipXecs/Roles,+Services+and+Processes


The next question is if it is supposed to be running, why does it come
up with this error?  I've sent profiles many times, restarted sipx and
even rebooted the server.

Any ideas?

Thanks!

Kyle


--
Kyle Haefner, M.S.
Communication Systems Programmer
Colorado State University
Fort Collins, CO
Phone:
 970-491-1012
Email:  kyle.haef...@colostate.edu
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-
No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.1913 / Virus Database: 2411/4948 - Release Date: 04/20/12 

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Re: [sipx-users] SharedAppearanceAgent does not match the software version (4.4.0).

[Tony Graziano]

have you compared the rpm versions?

On Fri, Apr 20, 2012 at 1:49 PM, Kyle Haefner wrote:

> Hi Users,
>
> I'm running Sipx 4.4 in a three node cluster with the most recent
> updates for 4.4.0- 2012-02-08EST09:10:08 ip-10-72-10-163.
>
>  In one of the redundant nodes when I get this:
>
> sudo sipxproc | grep SharedAppearanceAgent
>  "SharedAppearanceAgent"=>"Disabled",
>
> In the other redundant node i get this:
> sudo sipxproc | grep SharedAppearanceAgent
>  "SharedAppearanceAgent"=>"ConfigurationMismatch",
>
> sudo sipxproc -m SharedAppearanceAgent
> ["version.mismatch: software '4.4.0' != config ''"]
>
> So the question is, "Is SharedAppearanceAgent supposed to be running
> on redundant nodes"
>
> According to this it is supposed to be running.
> http://wiki.sipfoundry.org/display/sipXecs/Roles,+Services+and+Processes
>
>
> The next question is if it is supposed to be running, why does it come
> up with this error?  I've sent profiles many times, restarted sipx and
> even rebooted the server.
>
> Any ideas?
>
> Thanks!
>
> Kyle
>
>
> --
> Kyle Haefner, M.S.
> Communication Systems Programmer
> Colorado State University
> Fort Collins, CO
> Phone: 970-491-1012
> Email:  kyle.haef...@colostate.edu
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~

-- 
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net

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Blog: http://blog.myitdepartment.net
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[sipx-users] SharedAppearanceAgent does not match the software version (4.4.0).

[Kyle Haefner]

Hi Users,

I'm running Sipx 4.4 in a three node cluster with the most recent
updates for 4.4.0- 2012-02-08EST09:10:08 ip-10-72-10-163.

 In one of the redundant nodes when I get this:

sudo sipxproc | grep SharedAppearanceAgent
 "SharedAppearanceAgent"=>"Disabled",

In the other redundant node i get this:
sudo sipxproc | grep SharedAppearanceAgent
 "SharedAppearanceAgent"=>"ConfigurationMismatch",

sudo sipxproc -m SharedAppearanceAgent
["version.mismatch: software '4.4.0' != config ''"]

So the question is, "Is SharedAppearanceAgent supposed to be running
on redundant nodes"

According to this it is supposed to be running.
http://wiki.sipfoundry.org/display/sipXecs/Roles,+Services+and+Processes


The next question is if it is supposed to be running, why does it come
up with this error?  I've sent profiles many times, restarted sipx and
even rebooted the server.

Any ideas?

Thanks!

Kyle


-- 
Kyle Haefner, M.S.
Communication Systems Programmer
Colorado State University
Fort Collins, CO
Phone: 970-491-1012
Email:  kyle.haef...@colostate.edu
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Re: [sipx-users] Trunk to Trunk Transfer

[Tony Graziano]

Please test the following.

Calling from PSTN to AA and try to transfer to an extension that is not
forwarded.

You have to ensure your ITSP is sending the INVITE to port 5080 in order
for any transfers to succeed.

If not, what Gerald said... how to get around that is to create a dialplan
(56+10 digits sends 10 digits to a specific "other" gateway), which can
accomplish calls coming from one provider but the forward dialplan sends it
out through another. That should fix you.

On Fri, Apr 20, 2012 at 1:08 PM, Tommy Laino  wrote:

>
> Content-Type: text/plain;
>  charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <67729>
> Message-ID: <10891.4f919...@forum.sipfoundry.org>
>
>
>
> I have 3 IP trunks on my test system. I am trying to have an
> option from the auto attendant transfer to a cell phone. If
> i do it from a local or remote polycom phone it works fine.
> Once an external call comes into a trunk and it chooses the
> option the caller gets disconnected. Anything that I might
> be missing.
> --
> Tommy Laino
> Dome Technologies
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~

-- 
LAN/Telephony/Security and Control Systems Helpdesk:
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Re: [sipx-users] Trunk to Trunk Transfer

[Gerald Drouillard]

On 4/20/2012 1:08 PM, Tommy Laino wrote:
> Content-Type: text/plain;
>charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8<67729>
> Message-ID:<10891.4f919...@forum.sipfoundry.org>
>
>
>
> I have 3 IP trunks on my test system. I am trying to have an
> option from the auto attendant transfer to a cell phone. If
> i do it from a local or remote polycom phone it works fine.
> Once an external call comes into a trunk and it chooses the
> option the caller gets disconnected. Anything that I might
> be missing.
Your provider may not support hair pin transfers.  Who is your ITSP?

-- 
Regards
--
Gerald Drouillard
Technology Architect
Drouillard&  Associates, Inc.
http://www.Drouillard.biz

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[sipx-users] Trunk to Trunk Transfer

[Tommy Laino]

Content-Type: text/plain;
  charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <67729>
Message-ID: <10891.4f919...@forum.sipfoundry.org>



I have 3 IP trunks on my test system. I am trying to have an
option from the auto attendant transfer to a cell phone. If
i do it from a local or remote polycom phone it works fine.
Once an external call comes into a trunk and it chooses the
option the caller gets disconnected. Anything that I might
be missing.
-- 
Tommy Laino
Dome Technologies
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Re: [sipx-users] Fax Problems (Big Time)

[Tony Graziano]

did you do that? if so, did it fix it?

On Fri, Apr 20, 2012 at 11:53 AM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:

> Tony,
>
> ** **
>
> Thank you for the suggestion and for your help.
>
> ** **
>
> Rob
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Friday, April 20, 2012 11:00 AM
>
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] Fax Problems (Big Time)
>
> ** **
>
> Sorry, am not a audiocodes guru. I have documented the Patton PRi gateways
> for use with t.38 in the wiki. Perhaps someone else would be so kind as to
> do the same for an AC PRI gateway instead of keeping it a big secret and
> make sure it is also documented on the wiki.
>
> ** **
>
> Try:
>
> ** **
>
> Configuration > Protocol Configuration > Protocol Definition > SIP General
> Parameters
>
> Fax Signaling Method T.38 Relay
>
> Detect Fax on Answer Tone Initiate T.38 on Preamble
>
> SIP Transport Type UDP
>
> ** **
>
> I don't know what the other settings are for Detect Fax on Answer Tone,
> you really should address this before logging and tracing, because it
> should have fax set for t.38 on the Mediant FIRST.
>
> ** **
>
> For sanity, when you do find Nirvana, please post it here for someone to
> update the wiki if you don't have a wiki account.
>
> ** **
>
> On Fri, Apr 20, 2012 at 9:50 AM, Robert Schroeder <
> robert.schroe...@memberfirstmortgage.com> wrote:
>
> Tony:
>
>  
>
> I did find a Fax Signaling Method of T.38 Relay under the Coders And
> Profile Definitions – IP Profile Settings - Gateway Parameters – Fax
> Signaling Method.
>
>  
>
>  
>
>  
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Friday, April 20, 2012 9:29 AM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] Fax Problems (Big Time)
>
>  
>
> faxing into sipx only support t.38 and your audiocodes fax settings have
> g711u. you need to change it to t.38.
>
> On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder <
> robert.schroe...@memberfirstmortgage.com> wrote:
>
> Has anyone had problems with faxing into sipXecs? We are having huge
> problems with faxes being delivered to the users emails as zero page
> tiff’s. It is not an every fax occurrence however it is at a rate that I am
> no longer able to rely upon the faxing delivery capabilities of sipXecs. I
> am trying to determine if the problems is with the AudioCodes, PRI Provider
> or sipXecs.
>
>  
>
> Just wondering if there is a known issue. I am running (4.4.0-
> 2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits
> with an AudioCodes Mediant 1000 SIP Gateway.
>
>  
>
> Listed below is my current configuration settings on the AudioCodes!
>
>  
>
> *AudioCodes Information*
>
> Version ID: 6.00A.032.003
>
> DSP Type: 2
>
> DSP Software Version: 6008
>
> DSP Software Name: 624AE3
>
> Flash Version: 208
>
>  
>
> AudioCodes Configuration – Media Settings – Fax/Modem/CID Settings
>
>  
>
> *General Settings*
>
> Fax Transport Mode: RelayEnable
>
> Caller ID Transport Type: Mute
>
> Caller ID Type: Standard Bellcore
>
> V.21 Modem Transport Type: Disable
>
> V.22 Modem Transport Type: Disable
>
> V.23 Modem Transport Type: Disable
>
> V.32 Modem Transport Type: Disable
>
> V.34 Modem Transport Type: Enable Bypass
>
> Fax CNG Mode: Disable
>
> CNG Detector Mode: Disable
>
>  
>
> *Fax Relay Settings*
>
> Fax Relay Redundancy Depth: 0
>
> Fax Relay Enhanced Redundancy Depth: 4
>
> Fax Relay ECM Enable: Enable
>
> Fax Relay Max Rate (bps): 9600bps
>
>  
>
> *Bypass Settings*
>
> Fax/Modem Bypass Coder Type: G711Mulaw
>
> Fax/Modem Bypass Packing Factor: 1
>
> Fax Bypass Output Gain: 0
>
> Modem Bypass Output Gain: 0
>
>  
>
> Any suggestions from the techs would be very welcomed!
>
>  
>
> Thanks Everyone,
>
>  
>
> Rob Schroeder
>
>  
> --
>
>
> NOTICE: This electronic mail message and any content within it are intended 
> exclusively for the individual(s) or
>
> entities to which it is addressed. The message, together with any attachments 
> and all other content, may contain
>
> confidential and/or privileged information. Any unauthorized review, use, 
> print, save, copy, disclosure or distribution
>
> is strictly prohibited. If you have received this message in error, please 
> immediately advise the sender by reply email
> and delete all copies.
>
>
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Re: [sipx-users] Fax Problems (Big Time)

[Robert Schroeder]

Tony,
 
Thank you for the suggestion and for your help.
 
Rob
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 11:00 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
 
Sorry, am not a audiocodes guru. I have documented the Patton PRi gateways for 
use with t.38 in the wiki. Perhaps someone else would be so kind as to do the 
same for an AC PRI gateway instead of keeping it a big secret and make sure it 
is also documented on the wiki.
 
Try:
 
Configuration > Protocol Configuration > Protocol Definition > SIP General 
Parameters
Fax Signaling Method T.38 Relay
Detect Fax on Answer Tone Initiate T.38 on Preamble
SIP Transport Type UDP
 
I don't know what the other settings are for Detect Fax on Answer Tone, you 
really should address this before logging and tracing, because it should have 
fax set for t.38 on the Mediant FIRST.
 
For sanity, when you do find Nirvana, please post it here for someone to update 
the wiki if you don't have a wiki account.
 
On Fri, Apr 20, 2012 at 9:50 AM, Robert Schroeder 
 wrote:
Tony:
 
I did find a Fax Signaling Method of T.38 Relay under the Coders And Profile 
Definitions - IP Profile Settings - Gateway Parameters - Fax Signaling Method.
 
 
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 9:29 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
 
faxing into sipx only support t.38 and your audiocodes fax settings have g711u. 
you need to change it to t.38.
On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder 
 wrote:
Has anyone had problems with faxing into sipXecs? We are having huge problems 
with faxes being delivered to the users emails as zero page tiff's. It is not 
an every fax occurrence however it is at a rate that I am no longer able to 
rely upon the faxing delivery capabilities of sipXecs. I am trying to determine 
if the problems is with the AudioCodes, PRI Provider or sipXecs.
 
Just wondering if there is a known issue. I am running (4.4.0- 
2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with 
an AudioCodes Mediant 1000 SIP Gateway.
 
Listed below is my current configuration settings on the AudioCodes!
 
AudioCodes Information
Version ID: 6.00A.032.003
DSP Type: 2
DSP Software Version: 6008
DSP Software Name: 624AE3
Flash Version: 208
 
AudioCodes Configuration - Media Settings - Fax/Modem/CID Settings
 
General Settings
Fax Transport Mode: RelayEnable
Caller ID Transport Type: Mute
Caller ID Type: Standard Bellcore
V.21 Modem Transport Type: Disable
V.22 Modem Transport Type: Disable
V.23 Modem Transport Type: Disable
V.32 Modem Transport Type: Disable
V.34 Modem Transport Type: Enable Bypass
Fax CNG Mode: Disable
CNG Detector Mode: Disable
 
Fax Relay Settings
Fax Relay Redundancy Depth: 0
Fax Relay Enhanced Redundancy Depth: 4
Fax Relay ECM Enable: Enable
Fax Relay Max Rate (bps): 9600bps
 
Bypass Settings
Fax/Modem Bypass Coder Type: G711Mulaw
Fax/Modem Bypass Packing Factor: 1
Fax Bypass Output Gain: 0
Modem Bypass Output Gain: 0
 
Any suggestions from the techs would be very welcomed!
 
Thanks Everyone,
 
Rob Schroeder
 


NOTICE: This electronic mail message and any content within it are intended 
exclusively for the individual(s) or 
entities to which it is addressed. The message, together with any attachments 
and all other content, may contain
confidential and/or privileged information. Any unauthorized review, use, 
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is strictly prohibited. If you have received this message in error, please 
immediately advise the sender by reply email
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-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
 
Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
 
 


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entities to which it is addressed. The message, together with any attachments 
and all other content, may contain
confidential and/or privileged information. Any unauthorized review, use, 
print, save, copy, disclosure or distribution
is strictly prohibited. If you have received this message in error, please 
immediately advise the sender by reply email
and del

Re: [sipx-users] KIRK wiki guide inserted

[Geoff Van Brunt]

Thanks for this. We were thinking of doing it at one of our locations
but doing the evaluation is difficult without having the equipment.

Geoff Van Brunt



From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Philippe
Laurent
Sent: Thursday, April 19, 2012 3:40 PM
To: sipx-users
Subject: [sipx-users] KIRK wiki guide inserted

I've created a wiki guide within the Polycom section geared towards
setting up the KIRK server with SIPX. It's not written in a very
technical style, but it works nonetheless.

The two dozen KIRK phones that I've got get along well with SIPX.

Thanks for the opportunity to allow me to contribute.

Philippe
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Re: [sipx-users] Nat Problem

[Todd Hodgen]

+1

 

From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Joegen Baclor
Sent: Friday, April 20, 2012 2:29 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Nat Problem

 

I forgive all of you for not noticing that there is no ACK for the 200 OK.  
Forget RTP, it wont be there.  :-)

Check the jitsi side if it is respecting the Record-Route presented in 200 Ok.  
Take a wireshark capture from the PC too and send it together with the one from 
sipx.


On 04/20/2012 06:14 AM, Michael Picher wrote: 

or could be stun too if the media relay's just off in left field. 

 

i would setup for static outside address if you haven't.

 

mike

 

On Thu, Apr 19, 2012 at 4:59 PM, Tony Graziano  
wrote:

Most likely if that is the router sitting in front of sipx, yes. It's a 
prerequisite for media relay to function. 

 

1. Server behind nat has to be enabled.

2. Support remote workers needs to be enabled (sip trunking does not need to be 
enabled for remote workers, media relay does).

3. NAT for server needs to have access to a public IP address by getting it via 
STUN server or manually entering it.

4. Firewall in front of sipx needs to be able to do symmetrical outbound NAT 
(AON or FULL CONE NATE, at least for the outbound NAT of the sipx internal 
address) and have any SIP helper turned off.

 

If the two routers are adjacent and you can route (without NAT) you would 
simply do so and add the PC subnet to the intranets page in sipx ONLY IF it 
does not have to pass through NAT (i.e route or site to site vpn, etc.).

 

Your router is changing the ports and hence sipx is expecting the audio to come 
back on a port that isnt being sent by the router.

 

2012/4/19 Simon Brûlé 

So your saying the problem may come from the router I have (Linksys E2500) 
because it's not doing the symmetrical Nat so the RTP is getting lost?

2012/4/19 Tony Graziano 

If there is NAT between your PC and the sipx server then: 

 

1. The local firewall to your PC needs to have any SIP helper or Application 
Layer Gateway "turned off".

2. At your firewall where sipx is the  SIP helper or Application Layer Gateway 
" needs to be turned off AND the NAT type for the outbound NAT from the sipx 
server needs to be symmterical. Home brew and residential routers usually will 
not do this.

3. If you have an entry in your intranet subnets that includes the PC network 
number it must be listed ONLY IF IF DOES NOT GO THROUGH NAT.

 

One of these three (or a combination of them) is keeping RTP from flowing.

 

2012/4/19 Simon Brûlé 

I used the following command on the SipXecs server tcpdump -n -s 0 -i any -w 
filename.cap and then I transfer it on my computer so I could open it with my 
Wireshark and have a look at it. 

 

I joined the file to this e-mail so you could take a look and tell me what you 
think. This one is from a call I did from the softphone to the Hardphone where 
this one bugged like I described earlier.

 

Thanks.

 

2012/4/19 Simon Brûlé 

As i dig more and more in Wireshark i came to the conclusion that the Wireshark 
information that I just sent you is pretty much useless as I now see it. I will 
keep looking for some piece of information that could help. 

 

Thanks. 

 

2012/4/19 Simon Brûlé 

My Computer is connected in the Lan of the company and my E2500 is connected in 
this Lan too. My Server SipXecs and my hardphone are on the E2500. 

 

So there is an other router between my computer and the router that have my 
Server connected on it.

 

For the Wireshark part when I answer the phonecall I do from the softphone to 
my hardĥone those request are coming in until i close the call.

 

109 19.596310 192.168.175.22 192.168.175.136 SIP/SDP 1466 Status: 200 OK, with 
session description

 

Status-Code: 200

[resent packet : True]

[Suspected resend of frame:104]

[Request Frame : 57]

[Response Time (ms): 10950]

 

followed by this one:

 

110 19.598495 192.168.175.136 192.168.0.1 SIP 714 Request: ACK 
sip:3050@192.168.0.253:5060;transport=tcp

 

Request-Line: ACK sip:3050@192.168.0.253:5060;transport=tcp SIP/2.0

Method: ACK

Request-URI: sip:3050@192.168.0.253:5060;transport=tcp

[Resent Packet: False]

[Request Frame: 105]

[Response Time (ms): 512]

 

 

All those test have been done on Wireshark on the Computer with the Softphone 
on it. And the 192.168.0.253 that you see is the hardphone IP adresse. 

 

2012/4/19 Gerald Drouillard 

On 4/19/2012 3:25 PM, Simon Brûlé wrote: 

How can I do a capture with wireshark on the SipXecs server?

If you google a little you will find it. 





 

About the ALG you think that the other Router that give the DHCP to my Laptop 
and the Wan adresse of my router would have the Sip ALG activate?

That would be the only thing inbetween your softphone and the sipx server... 
right?
http://screenshots.portforward.com/Cisco/Linksys_E2500/Management.htm 






 

2012/4/19 Gerald Drouillar

Re: [sipx-users] Directed call pickup issue.

[Derrick Ding]

Thanks, Mike

Yes, I am setting up the environment for solving the issue: Aastra phone
doesn't accept SUBSCRIBE dialog or presence.  

 

Derrick

 



From: Michael Picher [mailto:mpic...@ezuce.com] 
Sent: Friday, April 20, 2012 10:54 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Directed call pickup issue.

 

I'd grab a capture of this and talk to your engineering folks.  This is
what we refer to as BLA functionality (bridged line appearance).

 

I would think this would be different from directed call pickup which is
*78ext .

 

If you can make this work and help get the Aastra template sorted and
new phones into templates I think you'll make a lot of happy folks who
would like another type of phone to use other than Polycom.

 

Also, on a testing basis I have a new Interop server on-line.  See Doc
here (the attached word doc on that page is formatted better):
http://wiki.sipfoundry.org/display/sipXecs/Interoperability+Testing

 

Thanks,

  Mike

On Fri, Apr 20, 2012 at 10:15 AM, Derrick Ding  wrote:

Hi All,

 

I use SipXecs as proxy server, and register an 2 Aastra phone and
Polycom phone.

I can monitor Polycom phone(phone number 203) from Aastra phone(phone
number 200). However when I do "directed call pickup", Aastra phone send
"INVITE *78203" to SipXecs, SipXec only responds 100 Trying, then
nothing happens.

 

What I did is:

1.  set BLF key on Aastra phone(ext 200) to monitor Polycom
phone(ext 203).
2.  Make a call from another phone to Polycom phone, Polycom phone
is ringing.
3.  It's observed that BLF key is flashing on Aastra phone.
4.  Press that key on Aastra phone

It's expected that Aastra phone will pickup the call.

It's observed that Aastra phone send "INVITE *78203" to SipXecs, SipXecs
responds 100 Trying, then nothing happens. Call is not pickup.

 

Do I need configure something on SipXecs?

 

Thanks for help.

Derrick

 


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-- 
Michael Picher, Director of Technical Services
eZuce, Inc.

300 Brickstone Square

Suite 201

Andover, MA. 01810

O.978-296-1005 X2015 
M.207-956-0262
@mpicher  
www.ezuce.com

 




There are 10 kinds of people in the world, those who understand binary
and those who don't.

 

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Re: [sipx-users] Fax Problems (Big Time)

[Tony Graziano]

Sorry, am not a audiocodes guru. I have documented the Patton PRi gateways
for use with t.38 in the wiki. Perhaps someone else would be so kind as to
do the same for an AC PRI gateway instead of keeping it a big secret and
make sure it is also documented on the wiki.

Try:

Configuration > Protocol Configuration > Protocol Definition > SIP General
Parameters
Fax Signaling Method T.38 Relay
Detect Fax on Answer Tone Initiate T.38 on Preamble
SIP Transport Type UDP

I don't know what the other settings are for Detect Fax on Answer Tone, you
really should address this before logging and tracing, because it should
have fax set for t.38 on the Mediant FIRST.

For sanity, when you do find Nirvana, please post it here for someone to
update the wiki if you don't have a wiki account.

On Fri, Apr 20, 2012 at 9:50 AM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:

> Tony:
>
> ** **
>
> I did find a Fax Signaling Method of T.38 Relay under the Coders And
> Profile Definitions – IP Profile Settings - Gateway Parameters – Fax
> Signaling Method.
>
> ** **
>
> ** **
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Friday, April 20, 2012 9:29 AM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] Fax Problems (Big Time)
>
> ** **
>
> faxing into sipx only support t.38 and your audiocodes fax settings have
> g711u. you need to change it to t.38.
>
> On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder <
> robert.schroe...@memberfirstmortgage.com> wrote:
>
> Has anyone had problems with faxing into sipXecs? We are having huge
> problems with faxes being delivered to the users emails as zero page
> tiff’s. It is not an every fax occurrence however it is at a rate that I am
> no longer able to rely upon the faxing delivery capabilities of sipXecs. I
> am trying to determine if the problems is with the AudioCodes, PRI Provider
> or sipXecs.
>
>  
>
> Just wondering if there is a known issue. I am running (4.4.0-
> 2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits
> with an AudioCodes Mediant 1000 SIP Gateway.
>
>  
>
> Listed below is my current configuration settings on the AudioCodes!
>
>  
>
> *AudioCodes Information*
>
> Version ID: 6.00A.032.003
>
> DSP Type: 2
>
> DSP Software Version: 6008
>
> DSP Software Name: 624AE3
>
> Flash Version: 208
>
>  
>
> AudioCodes Configuration – Media Settings – Fax/Modem/CID Settings
>
>  
>
> *General Settings*
>
> Fax Transport Mode: RelayEnable
>
> Caller ID Transport Type: Mute
>
> Caller ID Type: Standard Bellcore
>
> V.21 Modem Transport Type: Disable
>
> V.22 Modem Transport Type: Disable
>
> V.23 Modem Transport Type: Disable
>
> V.32 Modem Transport Type: Disable
>
> V.34 Modem Transport Type: Enable Bypass
>
> Fax CNG Mode: Disable
>
> CNG Detector Mode: Disable
>
>  
>
> *Fax Relay Settings*
>
> Fax Relay Redundancy Depth: 0
>
> Fax Relay Enhanced Redundancy Depth: 4
>
> Fax Relay ECM Enable: Enable
>
> Fax Relay Max Rate (bps): 9600bps
>
>  
>
> *Bypass Settings*
>
> Fax/Modem Bypass Coder Type: G711Mulaw
>
> Fax/Modem Bypass Packing Factor: 1
>
> Fax Bypass Output Gain: 0
>
> Modem Bypass Output Gain: 0
>
>  
>
> Any suggestions from the techs would be very welcomed!
>
>  
>
> Thanks Everyone,
>
>  
>
> Rob Schroeder
>
> ** **
> --
>
>
> NOTICE: This electronic mail message and any content within it are intended 
> exclusively for the individual(s) or
>
> entities to which it is addressed. The message, together with any attachments 
> and all other content, may contain
>
> confidential and/or privileged information. Any unauthorized review, use, 
> print, save, copy, disclosure or distribution
>
> is strictly prohibited. If you have received this message in error, please 
> immediately advise the sender by reply email
> and delete all copies.
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
> 
>
> ** **
>
> --
> ~~
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
> Fax: 434.465.6833
> ~~
> Linked-In Profile:
> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
> Ask about our Internet Fax services!
> ~~
>
> LAN/Telephony/Security and Control Systems Helpdesk:
>
> Telephone: 434.984.8426
>
> sip: helpd...@voice.myitdepartment.net
>
> ** **
>
> Helpdesk Customers: http://myhelp.myitdepartment.net
>
> Blog: http://blog.myitdepartment.net
>
>
> --
>
> NOTICE: This electronic mail message and any c

Re: [sipx-users] Directed call pickup issue.

[Michael Picher]

I'd grab a capture of this and talk to your engineering folks.  This is
what we refer to as BLA functionality (bridged line appearance).

I would think this would be different from directed call pickup which is
*78ext .

If you can make this work and help get the Aastra template sorted and new
phones into templates I think you'll make a lot of happy folks who would
like another type of phone to use other than Polycom.

Also, on a testing basis I have a new Interop server on-line.  See Doc here
(the attached word doc on that page is formatted better):
http://wiki.sipfoundry.org/display/sipXecs/Interoperability+Testing

Thanks,
  Mike

On Fri, Apr 20, 2012 at 10:15 AM, Derrick Ding  wrote:

>  Hi All,
>
> ** **
>
> I use SipXecs as proxy server, and register an 2 Aastra phone and Polycom
> phone.
>
> I can monitor Polycom phone(phone number 203) from Aastra phone(phone
> number 200). However when I do “directed call pickup”, Aastra phone send
> “INVITE *78203” to SipXecs, SipXec only responds 100 Trying, then nothing
> happens.
>
> ** **
>
> What I did is:
>
>1. set BLF key on Aastra phone(ext 200) to monitor Polycom phone(ext
>203).
>2. Make a call from another phone to Polycom phone, Polycom phone is
>ringing.
>3. It’s observed that BLF key is flashing on Aastra phone.
>4. Press that key on Aastra phone
>
> It’s expected that Aastra phone will pickup the call.
>
> It’s observed that Aastra phone send “INVITE *78203” to SipXecs, SipXecs
> responds 100 Trying, then nothing happens. Call is not pickup.
>
> ** **
>
> Do I need configure something on SipXecs?
>
> ** **
>
> Thanks for help.
>
> Derrick
>
> ** **
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



-- 
Michael Picher, Director of Technical Services
eZuce, Inc.

300 Brickstone Square

Suite 201

Andover, MA. 01810
O.978-296-1005 X2015
M.207-956-0262
@mpicher 
www.ezuce.com


There are 10 kinds of people in the world, those who understand binary and
those who don't.
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Re: [sipx-users] Bria for iphone/ipod touch registration

[Tony Graziano]

Please understand that sipxecs is meant for enterprise deployments. If it
were meant for smaller deployments as its market segment things would
probably be designed very differently, hence the prerequisite reading and
architectural understanding.

It is not aimed at users either, it's aimed at densely populated
environments with a well-rounded IT staff. It is not for everybody. With
some basic's learned though, it's easy enough to deploy if you fully
understand the concepts, but sidestepping fundementals does noone any
service.

On Fri, Apr 20, 2012 at 10:36 AM, Stiles Watson wrote:

>  Tony,
>
> The original intent of the email was to offer some constructive criticism
> and explain a little of my confusion. I never intended to make it personal.
> I realize that my "never let go of that bone" comment made it personal, and
> for that, I apologize.
>
> You guys have a superior product and I hope it continues to improve and
> continues to be used by more companies.
>
> Thank you, Tony, and everyone else, for your help.
>
> Stiles
>
>
>
> On 04/19/2012 01:31 PM, Tony Graziano wrote:
>
>
>
> On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson wrote:
>
>>  On 04/18/2012 06:15 PM, Tony Graziano wrote:
>>
>> You shouldn't invite violence. I have been known to clobber people with
>> iPads.
>>
>>  Why am I not surprised by this?
>>
>
>  Um that was a cartoon reference dude, you take things wayyy to
> seriously. Obviously you don't get out too much.
>
>>
>>
>>  I think the wiki could be clearer, but really I think you are the only
>> one to make the leap from Bria is bria is bria is bria. It ain't, but
>> sipfoundry doesn't manuafcture it either. Bria has undergone numerous name
>> changes in the last 3 or four years. I look for the names to change "yet
>> again", because why leave something the same?
>>
>>  I am not the only one who has been confused on this point. I have all
>> the emails from this list for the past year and a half or more. There have
>> been several people who have been confused about this. I know sipfoundry
>> does not manufacturer Bria, but you are saying that it is supported by your
>> software and therefore it is to your advantage to help your users/customers
>> use your product with whatever different versions exist of the third party
>> software you say you are interoperable with.
>>
>> Thanks for making my point about not taking a users view of things.
>>
>
>  Wrong, the wiki was changed by multiple editors to leave no chance for
> stiles.user error.. It clearly stated Bria 3.x, but the ASSUMPTION you had
> was all BRIA is BRIA. Whatever. It got changed to quell your issue. You
> didn't like the clarity of the wiki, so now it's idiot proffed (maybe) and
> you still want to complain about it.
>
>>
>> You guys do realize that the people that use your product are your
>> customers, right? I know no money changes hands, but you are creating this
>> for people to use, correct? You want companies big AND small to embrace
>> your product, correct? There are a number of potential users/customers who
>> have left your product because they could not get the help they thought
>> they needed.
>>
>
>  Its a very friendly environment. The wiki is there for a reason. There
> are many people who try to do things that LACK THE SKILLSET and/or FAIL TO
> READ OR UNDERSTAND what the best approach is to installing/configuring a
> system and then expect wayyy to much in having community members send them
> the wiki pages and explain the basic concepts. There are skills required as
> a prerequisite.
>
>>
>>
>>  Give people more information than they need? That's been done, numerous
>> times. When they can't follow directions the first few times, you have to
>> beat them over the head with an iPad and chant "DNS DNS DNS".
>>
>>  I love how you never let go of that bone! You are right, DNS is very
>> important and I have some things which still need to be configured. The
>> problem is that I do not think the DNS wiki page is perfectly clear and I
>> have been give contradictory info on this forum - again, its free support
>> offered by the sipX community which includes old hats, newbies, developers
>> and users so that happens. I was told that sipX sets up everything
>> correctly and that I do not need external DNS SRV records, but that was not
>> correct. It makes perfect sense now that I would need them.
>>
>
>  I think it was correct in the context or your related question(s) at
> that time, but clearly you want to pick yet another bone. Bringing this up
> out of context is just sour grapes and not really fair.
>
>>
>>
>>  Understand then fix your DNS and stop ranting to the masses who already
>> get that.
>>
>>  Ahh, just "Understand". It is easy to wave that wand when it is
>> something you are very familiar with. You might be one of those very gifted
>> people who instantly understands everything the first time they see it. I'm
>> happy for you - really.  However, on the whole, God has gifted different
>> people w

Re: [sipx-users] Bria for iphone/ipod touch registration

[Stiles Watson]

Tony,

The original intent of the email was to offer some constructive 
criticism and explain a little of my confusion. I never intended to make 
it personal. I realize that my "never let go of that bone" comment made 
it personal, and for that, I apologize.


You guys have a superior product and I hope it continues to improve and 
continues to be used by more companies.


Thank you, Tony, and everyone else, for your help.

Stiles


On 04/19/2012 01:31 PM, Tony Graziano wrote:



On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson > wrote:


On 04/18/2012 06:15 PM, Tony Graziano wrote:

You shouldn't invite violence. I have been known to clobber
people with iPads.

Why am I not surprised by this?


Um that was a cartoon reference dude, you take things wayyy to 
seriously. Obviously you don't get out too much.





I think the wiki could be clearer, but really I think you are the
only one to make the leap from Bria is bria is bria is bria. It
ain't, but sipfoundry doesn't manuafcture it either. Bria has
undergone numerous name changes in the last 3 or four years. I
look for the names to change "yet again", because why leave
something the same?

I am not the only one who has been confused on this point. I have
all the emails from this list for the past year and a half or
more. There have been several people who have been confused about
this. I know sipfoundry does not manufacturer Bria, but you are
saying that it is supported by your software and therefore it is
to your advantage to help your users/customers use your product
with whatever different versions exist of the third party software
you say you are interoperable with.

Thanks for making my point about not taking a users view of things.


Wrong, the wiki was changed by multiple editors to leave no chance for 
stiles.user error.. It clearly stated Bria 3.x, but the ASSUMPTION you 
had was all BRIA is BRIA. Whatever. It got changed to quell your 
issue. You didn't like the clarity of the wiki, so now it's idiot 
proffed (maybe) and you still want to complain about it.



You guys do realize that the people that use your product are your
customers, right? I know no money changes hands, but you are
creating this for people to use, correct? You want companies big
AND small to embrace your product, correct? There are a number of
potential users/customers who have left your product because they
could not get the help they thought they needed.


Its a very friendly environment. The wiki is there for a reason. There 
are many people who try to do things that LACK THE SKILLSET and/or 
FAIL TO READ OR UNDERSTAND what the best approach is to 
installing/configuring a system and then expect wayyy to much in 
having community members send them the wiki pages and explain the 
basic concepts. There are skills required as a prerequisite.





Give people more information than they need? That's been done,
numerous times. When they can't follow directions the first few
times, you have to beat them over the head with an iPad and chant
"DNS DNS DNS".

I love how you never let go of that bone! You are right, DNS is
very important and I have some things which still need to be
configured. The problem is that I do not think the DNS wiki page
is perfectly clear and I have been give contradictory info on this
forum - again, its free support offered by the sipX community
which includes old hats, newbies, developers and users so that
happens. I was told that sipX sets up everything correctly and
that I do not need external DNS SRV records, but that was not
correct. It makes perfect sense now that I would need them.


I think it was correct in the context or your related question(s) at 
that time, but clearly you want to pick yet another bone. Bringing 
this up out of context is just sour grapes and not really fair.





Understand then fix your DNS and stop ranting to the masses who
already get that.

Ahh, just "Understand". It is easy to wave that wand when it is
something you are very familiar with. You might be one of those
very gifted people who instantly understands everything the first
time they see it. I'm happy for you - really.  However, on the
whole, God has gifted different people with different abilities.
The thing which is easy for one, may very very difficult for another.

DNS, DNS, DNS! What beautiful words! On to DNS!


Dude, the whole thing is: DNS, Networking, firewalls, all play a part 
in any VOIP deployment. The wiki article on DNS is written for 
administrators. Why? Because DNS is VERY important to any environment 
and easy to break. It's not in the best interest of any project 
focusing on SIP to teach DNS 101 to admins who don't understand how 
DNS works. There is no real easy way to explain DNS. It's not fair to 
tech network 101 courses. You 

[sipx-users] Directed call pickup issue.

[Derrick Ding]

Hi All,

 

I use SipXecs as proxy server, and register an 2 Aastra phone and
Polycom phone.

I can monitor Polycom phone(phone number 203) from Aastra phone(phone
number 200). However when I do "directed call pickup", Aastra phone send
"INVITE *78203" to SipXecs, SipXec only responds 100 Trying, then
nothing happens.

 

What I did is:

1.  set BLF key on Aastra phone(ext 200) to monitor Polycom
phone(ext 203).
2.  Make a call from another phone to Polycom phone, Polycom phone
is ringing.
3.  It's observed that BLF key is flashing on Aastra phone.
4.  Press that key on Aastra phone

It's expected that Aastra phone will pickup the call.

It's observed that Aastra phone send "INVITE *78203" to SipXecs, SipXecs
responds 100 Trying, then nothing happens. Call is not pickup.

 

Do I need configure something on SipXecs?

 

Thanks for help.

Derrick

 

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Re: [sipx-users] Fax Problems (Big Time)

[Robert Schroeder]

Tony:
 
I did find a Fax Signaling Method of T.38 Relay under the Coders And Profile 
Definitions - IP Profile Settings - Gateway Parameters - Fax Signaling Method.
 
 
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 9:29 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
 
faxing into sipx only support t.38 and your audiocodes fax settings have g711u. 
you need to change it to t.38.
On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder 
 wrote:
Has anyone had problems with faxing into sipXecs? We are having huge problems 
with faxes being delivered to the users emails as zero page tiff's. It is not 
an every fax occurrence however it is at a rate that I am no longer able to 
rely upon the faxing delivery capabilities of sipXecs. I am trying to determine 
if the problems is with the AudioCodes, PRI Provider or sipXecs.
 
Just wondering if there is a known issue. I am running (4.4.0- 
2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with 
an AudioCodes Mediant 1000 SIP Gateway.
 
Listed below is my current configuration settings on the AudioCodes!
 
AudioCodes Information
Version ID: 6.00A.032.003
DSP Type: 2
DSP Software Version: 6008
DSP Software Name: 624AE3
Flash Version: 208
 
AudioCodes Configuration - Media Settings - Fax/Modem/CID Settings
 
General Settings
Fax Transport Mode: RelayEnable
Caller ID Transport Type: Mute
Caller ID Type: Standard Bellcore
V.21 Modem Transport Type: Disable
V.22 Modem Transport Type: Disable
V.23 Modem Transport Type: Disable
V.32 Modem Transport Type: Disable
V.34 Modem Transport Type: Enable Bypass
Fax CNG Mode: Disable
CNG Detector Mode: Disable
 
Fax Relay Settings
Fax Relay Redundancy Depth: 0
Fax Relay Enhanced Redundancy Depth: 4
Fax Relay ECM Enable: Enable
Fax Relay Max Rate (bps): 9600bps
 
Bypass Settings
Fax/Modem Bypass Coder Type: G711Mulaw
Fax/Modem Bypass Packing Factor: 1
Fax Bypass Output Gain: 0
Modem Bypass Output Gain: 0
 
Any suggestions from the techs would be very welcomed!
 
Thanks Everyone,
 
Rob Schroeder
 


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sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
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Re: [sipx-users] Fax Problems (Big Time)

[Robert Schroeder]

Tony,
 
Thank you for the suggestion. Are you referring to the G711Mulaw setting under 
Bypass Settings? The only other option is G711Alaw_64.
 
 
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 9:29 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
 
faxing into sipx only support t.38 and your audiocodes fax settings have g711u. 
you need to change it to t.38.
On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder 
 wrote:
Has anyone had problems with faxing into sipXecs? We are having huge problems 
with faxes being delivered to the users emails as zero page tiff's. It is not 
an every fax occurrence however it is at a rate that I am no longer able to 
rely upon the faxing delivery capabilities of sipXecs. I am trying to determine 
if the problems is with the AudioCodes, PRI Provider or sipXecs.
 
Just wondering if there is a known issue. I am running (4.4.0- 
2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with 
an AudioCodes Mediant 1000 SIP Gateway.
 
Listed below is my current configuration settings on the AudioCodes!
 
AudioCodes Information
Version ID: 6.00A.032.003
DSP Type: 2
DSP Software Version: 6008
DSP Software Name: 624AE3
Flash Version: 208
 
AudioCodes Configuration - Media Settings - Fax/Modem/CID Settings
 
General Settings
Fax Transport Mode: RelayEnable
Caller ID Transport Type: Mute
Caller ID Type: Standard Bellcore
V.21 Modem Transport Type: Disable
V.22 Modem Transport Type: Disable
V.23 Modem Transport Type: Disable
V.32 Modem Transport Type: Disable
V.34 Modem Transport Type: Enable Bypass
Fax CNG Mode: Disable
CNG Detector Mode: Disable
 
Fax Relay Settings
Fax Relay Redundancy Depth: 0
Fax Relay Enhanced Redundancy Depth: 4
Fax Relay ECM Enable: Enable
Fax Relay Max Rate (bps): 9600bps
 
Bypass Settings
Fax/Modem Bypass Coder Type: G711Mulaw
Fax/Modem Bypass Packing Factor: 1
Fax Bypass Output Gain: 0
Modem Bypass Output Gain: 0
 
Any suggestions from the techs would be very welcomed!
 
Thanks Everyone,
 
Rob Schroeder
 


NOTICE: This electronic mail message and any content within it are intended 
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confidential and/or privileged information. Any unauthorized review, use, 
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-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
 
Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net


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Re: [sipx-users] Fax Problems (Big Time)

[George Niculae]

On Fri, Apr 20, 2012 at 4:38 PM, Robert Schroeder
 wrote:
> George,
>
It's Voicemail and AA, however pay attention to Tony's reply: sipx
only support t.38

George
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Re: [sipx-users] Fax Problems (Big Time)

[Tony Graziano]

faxing into sipx only support t.38 and your audiocodes fax settings have
g711u. you need to change it to t.38.

On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:

> Has anyone had problems with faxing into sipXecs? We are having huge
> problems with faxes being delivered to the users emails as zero page
> tiff’s. It is not an every fax occurrence however it is at a rate that I am
> no longer able to rely upon the faxing delivery capabilities of sipXecs. I
> am trying to determine if the problems is with the AudioCodes, PRI Provider
> or sipXecs.
>
> ** **
>
> Just wondering if there is a known issue. I am running (4.4.0-
> 2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits
> with an AudioCodes Mediant 1000 SIP Gateway.
>
> ** **
>
> Listed below is my current configuration settings on the AudioCodes!
>
> ** **
>
> *AudioCodes Information*
>
> Version ID: 6.00A.032.003
>
> DSP Type: 2
>
> DSP Software Version: 6008
>
> DSP Software Name: 624AE3
>
> Flash Version: 208
>
> ** **
>
> AudioCodes Configuration – Media Settings – Fax/Modem/CID Settings
>
> ** **
>
> *General Settings*
>
> Fax Transport Mode: RelayEnable
>
> Caller ID Transport Type: Mute
>
> Caller ID Type: Standard Bellcore
>
> V.21 Modem Transport Type: Disable
>
> V.22 Modem Transport Type: Disable
>
> V.23 Modem Transport Type: Disable
>
> V.32 Modem Transport Type: Disable
>
> V.34 Modem Transport Type: Enable Bypass
>
> Fax CNG Mode: Disable
>
> CNG Detector Mode: Disable
>
> ** **
>
> *Fax Relay Settings*
>
> Fax Relay Redundancy Depth: 0
>
> Fax Relay Enhanced Redundancy Depth: 4
>
> Fax Relay ECM Enable: Enable
>
> Fax Relay Max Rate (bps): 9600bps
>
> ** **
>
> *Bypass Settings*
>
> Fax/Modem Bypass Coder Type: G711Mulaw
>
> Fax/Modem Bypass Packing Factor: 1
>
> Fax Bypass Output Gain: 0
>
> Modem Bypass Output Gain: 0
>
> ** **
>
> Any suggestions from the techs would be very welcomed!
>
> ** **
>
> Thanks Everyone,
>
> ** **
>
> Rob Schroeder
>
> 
>
>
> --
>
> NOTICE: This electronic mail message and any content within it are intended 
> exclusively for the individual(s) or
>
> entities to which it is addressed. The message, together with any attachments 
> and all other content, may contain
>
> confidential and/or privileged information. Any unauthorized review, use, 
> print, save, copy, disclosure or distribution
>
> is strictly prohibited. If you have received this message in error, please 
> immediately advise the sender by reply email
> and delete all copies.
>
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~

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Re: [sipx-users] Fax Problems (Big Time)

[George Niculae]

On Fri, Apr 20, 2012 at 4:17 PM, Robert Schroeder
 wrote:
> Has anyone had problems with faxing into sipXecs? We are having huge
> problems with faxes being delivered to the users emails as zero page tiff’s.
> It is not an every fax occurrence however it is at a rate that I am no
> longer able to rely upon the faxing delivery capabilities of sipXecs. I am
> trying to determine if the problems is with the AudioCodes, PRI Provider or
> sipXecs.
>

Please attach sipxivr.log. Would be even better if log level set on debug

George
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[sipx-users] Fax Problems (Big Time)

[Robert Schroeder]

Has anyone had problems with faxing into sipXecs? We are having huge problems 
with faxes being delivered to the users emails as zero page tiff's. It is not 
an every fax occurrence however it is at a rate that I am no longer able to 
rely upon the faxing delivery capabilities of sipXecs. I am trying to determine 
if the problems is with the AudioCodes, PRI Provider or sipXecs.
 
Just wondering if there is a known issue. I am running (4.4.0- 
2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with 
an AudioCodes Mediant 1000 SIP Gateway.
 
Listed below is my current configuration settings on the AudioCodes!
 
AudioCodes Information
Version ID: 6.00A.032.003
DSP Type: 2
DSP Software Version: 6008
DSP Software Name: 624AE3
Flash Version: 208
 
AudioCodes Configuration - Media Settings - Fax/Modem/CID Settings
 
General Settings
Fax Transport Mode: RelayEnable
Caller ID Transport Type: Mute
Caller ID Type: Standard Bellcore
V.21 Modem Transport Type: Disable
V.22 Modem Transport Type: Disable
V.23 Modem Transport Type: Disable
V.32 Modem Transport Type: Disable
V.34 Modem Transport Type: Enable Bypass
Fax CNG Mode: Disable
CNG Detector Mode: Disable
 
Fax Relay Settings
Fax Relay Redundancy Depth: 0
Fax Relay Enhanced Redundancy Depth: 4
Fax Relay ECM Enable: Enable
Fax Relay Max Rate (bps): 9600bps
 
Bypass Settings
Fax/Modem Bypass Coder Type: G711Mulaw
Fax/Modem Bypass Packing Factor: 1
Fax Bypass Output Gain: 0
Modem Bypass Output Gain: 0
 
Any suggestions from the techs would be very welcomed!
 
Thanks Everyone,
 
Rob Schroeder
 


NOTICE: This electronic mail message and any content within it are intended 
exclusively for the individual(s) or 
entities to which it is addressed. The message, together with any attachments 
and all other content, may contain
confidential and/or privileged information. Any unauthorized review, use, 
print, save, copy, disclosure or distribution
is strictly prohibited. If you have received this message in error, please 
immediately advise the sender by reply email
and delete all copies.
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