Re: [sipx-users] Grandstream GXW4104

2012-04-23 Thread Tony Graziano 
Really the best thing you can do is put your log with sipx (proxy) to
debug, and grab whatever best level of detail/logging you can from your
gateway. I don't think this happens with others and people probably arent
answering you because either it doesnt work well for them or the MFR simply
doesnt provide an adequate sip stack or support.

If you see something in the logs, post it here, but you need to discern
WHERE the BYE is coming from. Since the RTP is established between the UA
(phone) and the gateway, sipx is mostly out of the picture except recording
the BYE to cut the CRD record. This is why it is important to use a good
network infrastructure along with the gateway and handset, of course.

There are a couple of easy gateways to use: AudioCodes and Patton. For less
detailed configuration options and ease of configuration a lot of people
choose Audiocodes. (not me).

Good luck.

2012/4/23 Nitin Mirchandani 

>  I have one suggestion for you - Dont use Grandstream. I dont know which
> stack they use - But be it gateway or phone - Its simply unstable (gave up
> trying)
>
> --
> Date: Mon, 23 Apr 2012 11:54:14 -0700
> From: branderso...@msn.com
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Grandstream GXW4104
>
>
> Could Problem number two be caused by incorrect Refresher, or timer
> settings?  If so, what should they be?
>
> On the gateway:
>
> *Session Expiration: * (in seconds. default 180 seconds) *
> Min-SE: *   (in seconds. default and minimum 90 seconds) *
> Caller Request Timer: *   Yes No (Request for timer when making
> outbound calls)
> *Callee Request Timer: *   Yes No (When caller supports timer but did
> not request one) *
> Force Timer: *   Yes No (Use timer even when remote party does not
> support)
> *UAC Specify Refresher: *   UAC   UAS Omit (Recommended) *
> UAS Specify Refresher: *   UAC   UAS (When UAC did not specify refresher
> tag)
>
>
>
> -Bryan Anderson
>
>
>
> On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson wrote:
>
> I have been having issues with a new Grandstream GXW4104 fxo gateway and
> was wondering if anyone could help.
>
> We have 4 pstn lines from qwest going into the gateway.   All calls go to
> an Auto Attendant when answered.
>
> the two problems we have experienced are:
>
> 1) After about 1-1.5 hours the call hit the Auto Attendant but wont
> transfer out.  Some dials and extension they just get dead air.  (this is
> fixed by rebooting the gateway.)
>
> 2) The external uses (either some one who called it, or some one we have
> called) stop hearing audio, but we can still here them. This happens
> anywhere from 1-10 minutes into the call.
>
> sipXecs (4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1)
>
> Grandstream firmware: Program--1.3.4.13Loader--1.1.3.4Boot--1.1.3.2
>
> The phones are 1 Polycom IP 550 and 7 Polycom SoundPoint IP 331
>
> -Bryan Anderson
>
>
>
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>
>
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Re: [sipx-users] Grandstream GXW4104

2012-04-23 Thread Nitin Mirchandani 

I have one suggestion for you - Dont use Grandstream. I dont know which stack 
they use - But be it gateway or phone - Its simply unstable (gave up trying)
 Date: Mon, 23 Apr 2012 11:54:14 -0700
From: branderso...@msn.com
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Grandstream GXW4104

Could Problem number two be caused by incorrect Refresher, or timer settings?  
If so, what should they be?

On the gateway:

Session Expiration:  
  (in seconds. default 180 seconds)
  
  
  

Min-SE: 
  
  (in seconds. default and minimum 90 seconds)
  
  
  

Caller Request Timer: 
  
  Yes 
  No (Request for timer when making outbound calls)
Callee Request Timer: 
  
  Yes 
  No (When caller supports timer but did not request one)
  

  

Force Timer: 
  
  Yes 
  No (Use timer even when remote party does not support)
  

  

UAC Specify Refresher: 
  
  UAC   
  UAS 
  Omit (Recommended)
   
 
  

UAS Specify Refresher: 
  
  UAC   
  UAS (When UAC did not specify refresher tag)


-Bryan Anderson




On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson  wrote:

I have been having issues with a new Grandstream GXW4104 fxo gateway and was 
wondering if anyone could help.


We have 4 pstn lines from qwest going into the gateway.   All calls go to an 
Auto Attendant when answered.


the two problems we have experienced are:

1) After about 1-1.5 hours the call hit the Auto Attendant but wont transfer 
out.  Some dials and extension they just get dead air.  (this is fixed by 
rebooting the gateway.)



2) The external uses (either some one who called it, or some one we have 
called) stop hearing audio, but we can still here them. This happens anywhere 
from 1-10 minutes into the call.

sipXecs
(4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1)

Grandstream firmware: Program--1.3.4.13Loader--1.1.3.4Boot--1.1.3.2

The phones are 1 Polycom IP 550 and 7 Polycom SoundPoint IP 331

-Bryan Anderson





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Re: [sipx-users] Grandstream GXW4104

2012-04-23 Thread Bryan Anderson 
Could Problem number two be caused by incorrect Refresher, or timer
settings?  If so, what should they be?

On the gateway:

*Session Expiration: * (in seconds. default 180 seconds) *
Min-SE: *   (in seconds. default and minimum 90 seconds) *
Caller Request Timer: *   Yes No (Request for timer when making
outbound calls)
*Callee Request Timer: *   Yes No (When caller supports timer but did
not request one) *
Force Timer: *   Yes No (Use timer even when remote party does not
support)
*UAC Specify Refresher: *   UAC   UAS Omit (Recommended) *
UAS Specify Refresher: *   UAC   UAS (When UAC did not specify refresher
tag)



-Bryan Anderson



On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson wrote:

> I have been having issues with a new Grandstream GXW4104 fxo gateway and
> was wondering if anyone could help.
>
> We have 4 pstn lines from qwest going into the gateway.   All calls go to
> an Auto Attendant when answered.
>
> the two problems we have experienced are:
>
> 1) After about 1-1.5 hours the call hit the Auto Attendant but wont
> transfer out.  Some dials and extension they just get dead air.  (this is
> fixed by rebooting the gateway.)
>
> 2) The external uses (either some one who called it, or some one we have
> called) stop hearing audio, but we can still here them. This happens
> anywhere from 1-10 minutes into the call.
>
> sipXecs (4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1)
>
> Grandstream firmware: Program--1.3.4.13Loader--1.1.3.4Boot--1.1.3.2
>
> The phones are 1 Polycom IP 550 and 7 Polycom SoundPoint IP 331
>
> -Bryan Anderson
>
>
>
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> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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Re: [sipx-users] voip.ms

2012-04-23 Thread Gerald Drouillard 
On 4/23/2012 9:45 AM, Kumaran wrote:
> Hi All,
> Whether Voip.ms supports t.38 codec so that I can assign a DID number
> to user fax extension?
>
> Regards,
> Kumaran T
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For low cost pay per usage plans:

I had a some success with callcentric.com.  See:
http://www.drouillard.biz/blog/sipxecs-hylafax-and-t38modem/

Some that I haven't tried yet that may work:
http://www.voicepulse.com/
http://www.t38faxing.com/

For large installs where you have over $200/month in services you may 
want to consider:
http://www.voipinnovations.com/


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

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Re: [sipx-users] Generate CSR Question

2012-04-23 Thread Josh Patten 
In addition to doing this you should also add these chain certificates to
the web certificate you upload to sipxconfig. This is so that the full
certificate chain can be loaded and presented to browsers. To do this
simply add the intermediate certificate chains to the beginning of the web
SSL cert like so:

-BEGIN CERTIFICATE-
intermediate-cert-text-goes-here
-END CERTIFICATE-
-BEGIN CERTIFICATE-
2nd-intermediate-cert-text-goes-here
-END CERTIFICATE-
-BEGIN CERTIFICATE-
3rd-intermediate-cert-text-goes-here
-END CERTIFICATE-
-BEGIN CERTIFICATE-
SSL-cert-text-goes-here
-END CERTIFICATE-

On Fri, Apr 20, 2012 at 4:09 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
>
> ** **
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Re: [sipx-users] voip.ms

2012-04-23 Thread Michael Picher 
does not support t.38

On Mon, Apr 23, 2012 at 9:45 AM, Kumaran <
thiru.venkateshwa...@ttplservices.com> wrote:

> Hi All,
>   Whether Voip.ms supports t.38 codec so that I can assign a DID number
> to user fax extension?
>
> Regards,
> Kumaran T
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>



-- 
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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


There are 10 kinds of people in the world, those who understand binary and
those who don't.
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[sipx-users] voip.ms

2012-04-23 Thread Kumaran 
Hi All,
   Whether Voip.ms supports t.38 codec so that I can assign a DID number 
to user fax extension?

Regards,
Kumaran T
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