Could you please advise how I can setup an unmaged gateway? If we do that, do 
we 
just directly connect to the bandwidth's router instead of their voip box? 
Please advise.

Thanks and have a nice day,

-SrinvasaRao Seelam




________________________________
From: Tony Graziano <tgrazi...@myitdepartment.net>
To: Discussion list for users of sipXecs software 
<sipx-users@list.sipfoundry.org>
Sent: Sun, January 9, 2011 7:04:40 PM
Subject: Re: [sipx-users] auto-attendant messages are played real fast

Ugh. You are using one of those pesky edgemarc boxes from them. You should be 
simply doing an unmanaged gateway from sipx to their box. Your should ask them 
why this is happening. Every one I've ever touched I de-installed and run a 
firewall in front of it.  


Here is how to do a call trace.

http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer


A properly configured firewall (like pfSense) can keep the edgemarc from being 
needed at all, but I've had many reports of audio issue with bandwidth.com 
since 
the start of business on Monday (Jan. 3rd). Is this something that "was 
working" 
but started recently also? I suspect they have made some core changes on their 
network, because we started to see broken 2-way audio, etc., also.


On Sun, Jan 9, 2011 at 5:36 PM, srinivasa rao <ssv...@yahoo.com> wrote:

We are using Bandwidth.com's equipment. Bandwidth.com gave us a WAN device and 
VoIP device. 

>
>T1 connected to a WAN device, other end of WAN device is connected to the VoIP 
>device. Then the VoIP device is connected to the SIPX box. 
>
>I will be glad to run a call trace in sipx. Can you point me to some 
>documentation that tells me how to run call trace in SIPX?
>
>
>Thanks and have a nice day,
>
>-SrinivasaRao Seelam 
>
> 
>
>
>
________________________________

>From: Tony Graziano <tgrazi...@myitdepartment.net>
>
>To: Discussion list for users of sipXecs software 
><sipx-users@list.sipfoundry.org>
>Sent: Sun, January 9, 2011 5:13:38 PM 
>
>Subject: Re: [sipx-users] auto-attendant messages are played real fast
>
>
>
>
>
>On Sun, Jan 9, 2011 at 11:23 AM, srinivasa rao <ssv...@yahoo.com> wrote:
>
>1) The phones are Polycom IP 430
>>
ok 
2) Firmware is 3.2.3.1734 
>
(this firmware is known to have issues, but the calls are not terminating at 
the 
phone) 
3) When I dial 0 on a IP 430 phone located in our pbx/network to here these auto
>attendent messages, they are crystal clear.
>
good 
4) However, when I call the main number from outside, the very same messages are
>played real fast.
>
Do you have more than one gateway/provider to test against? This would really 
indicate an issue with FS (in sipx), but I have only heard of a couple of 
 issues when the FS system was using "Play_fsv" in freeswitch. Why that is 
happening is anyone's guess. Perhaps a call trace could shed some light on what 
is different with the gateway connected call.

>Thanks and have a nice day,
>
>-srinivasarao seelam  
>
>
>
>
>----- Original Message ----
>From: Tony Graziano <tgrazi...@myitdepartment.net>
>
>To: sipx-users@list.sipfoundry.org
>Sent: Sat, January 8, 2011 7:00:37 PM
>Subject: Re: [sipx-users] auto-attendant messages are played real fast
>
>What phone and firmware, this was never answered.
>============================
>Tony Graziano, Manager
>Telephone: 434.984.8430
>Fax: 434.984.8431
>
>Email: tgrazi...@myitdepartment.net
>
>LAN/Telephony/Security and Control Systems Helpdesk:
>Telephone: 434.984.8426
>Fax: 434.984.8427
>
>Helpdesk Contract Customers:
>http://www.myitdepartment.net/gethelp/
>
>----- Original Message -----
>From: sipx-users-boun...@list.sipfoundry.org
><sipx-users-boun...@list.sipfoundry.org>
>To: Discussion list for users of sipXecs software
><sipx-users@list.sipfoundry.org>
>Sent: Sat Jan 08 18:50:00 2011
>Subject: Re: [sipx-users] auto-attendant messages are played real fast
>
>I tried that; basically they exactly look the same: I gave the following
>outputs
>
>
>
>[root@sipx1 prompts]# file autoattendant.wav
>autoattendant.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
>bit,
>mono 8000 Hz
>
>[root@sipx1 prompts]# file main.wav
>main.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono
>8000 Hz
>[root@sipx1 prompts]#
>
>
>In the above output, autoattendant.wav came with the installation. And the
>main.wav file is the one I uploaded.
>
>Like I mentioned, it was working great; the problem just started a few weeks
>ago. Also, I noticed that any auto attendant message that is upload is
>getting
>played very fast.
>
>Kindly advise what should I try next.
>
>Thanks and have a nice day,
>
>SrinivasaRao Seelam
>
>
>----- Original Message ----
>From: Douglas Hubler <dhub...@ezuce.com>
>To: Discussion list for users of sipXecs software
><sipx-users@list.sipfoundry.org>
>Sent: Wed, January 5, 2011 11:29:20 PM
>Subject: Re: [sipx-users] auto-attendant messages are played real fast
>
>On Wed, Jan 5, 2011 at 4:40 PM, srinivasa rao <ssv...@yahoo.com> wrote:
>> We have auto attendants setup in our sipx setup. For the last few weeks,
>> these
>> auto attendant messages are played very very fast. For example, our main
>>message
>>
>> used to take 40 seconds to complete it, and it takes very less time to
>complete
>> it.
>>
>> However, when I play the auto attendant messages manually in the sipx
>> administration page, the auto attendent messages are played correctly.
>>
>> This is a non profit agency, and I would really appreciate your feed back
>> in
>> resolving the issue.
>
>Run this command on the *.wav files
>  file myfile.wav
>
>Example:
>
>file tryagain.wav
>  tryagain.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM,
>16 bit, mono 8000 Hz
>
>
>it will show you the format details.  Check the bitrate and encoding
>on a file that come with sipxecs.
>_______________________________________________
>sipx-users mailing list
>sipx-users@list.sipfoundry.org
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>
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>
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>sipx-users@list.sipfoundry.org
>List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
>
>
>_______________________________________________
>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: 434.326.5325

Email: tgrazi...@myitdepartment.net

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

Helpdesk Contract Customers:
http://support.myitdepartment.net 

Blog:
http://blog.myitdepartment.net


Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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