you would need to use an sbc to do that. Alternatively, you can have a properly configured firewall to do the nat translation and setup a standard sip trunk in sipx (which is what I usually end up doing). Both of these methods would bypass the bandwidth.com hardware device.
On Mon, Jan 31, 2011 at 10:33 PM, srinivasa rao <[email protected]> wrote: > 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 <[email protected]> > *To:* Discussion list for users of sipXecs software < > [email protected]> > *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 <[email protected]> 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 <[email protected]> >> *To:* Discussion list for users of sipXecs software < >> [email protected]> >> *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 <[email protected]> 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 <[email protected]> >>> To: [email protected] >>> 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: [email protected] >>> >>> 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: [email protected] >>> <[email protected]> >>> To: Discussion list for users of sipXecs software >>> <[email protected]> >>> 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 <[email protected]> >>> To: Discussion list for users of sipXecs software >>> <[email protected]> >>> 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 <[email protected]> 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 >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > > <http://support.myitdepartment.net/>Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <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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