[Freeswitch-users] Grandstream gateways
Nandy Dagondon
nandy1925 at gmail.com
Fri Apr 23 02:16:24 PDT 2010
@kendalll: CallerID number is fine for PSTN application. what model do you
have? any experience w/ their 24 port FXS?
i'm really looking for 24 ports FXS for PSTN application. aside from
audiocodes, any brand can you recommend? the FXS must generate 12/16khz
pulse metering signal. tks.
-nandy
On Thu, Dec 10, 2009 at 4:12 AM, Kendall Stauffer <ken at ksac.com> wrote:
> Yes. I have one if anybody wants it, would let it go cheap.
>
>
>
> Works fine, but caller id is only the number, not the name part. Other
> than that works fine with astersik
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Toro
> *Sent:* Wednesday, December 09, 2009 3:05 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Grandstream gateways
>
>
>
> I had hear about Welltech (http://www.welltech.com/default.aspx) gateways
> but I don't have any experience with them.
>
>
>
> Someone know ?, any experience...
>
>
>
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
> --- On *Wed, 11/25/09, Milena <testeador01 at gmail.com>* wrote:
>
>
> From: Milena <testeador01 at gmail.com>
> Subject: Re: [Freeswitch-users] Grandstream gateways
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, November 25, 2009, 4:00 PM
>
> Hello,
>
>
>
> Samuel: We also have some GXW4104 gateways, in small production/testing
> environments; our caller id works fine and none of them has failed in over a
> year of being used. The thing that i dislike about the GXW series is that it
> has no support for early media.
>
>
>
> Everyone: What FXO devices do you currently use / recommend?
>
>
>
> 2009/11/25 Chris Chen <chris.chen2004 at gmail.com<http://mc/compose?to=chris.chen2004@gmail.com>
> >
>
> You haven't really put it into production for more than one year. The issue
> with GXW4108 is that after some time, say a couple of months, either all FXO
> ports not working, or worse, some FXO ports not working, but after power
> recycling, they will come back to work for some time until on strike again
> at some time you have no control.
>
> This had been reported for a couple of years without improvement. Go google
> search you will find out, this has happened to many GXW4108 users.
>
>
> On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti <samuelmukoti at gmail.com<http://mc/compose?to=samuelmukoti@gmail.com>>
> wrote:
>
> Thank you for those tips,
>
> I do have some small setups using gxw4108 they work or, except CID
> doesn't seem to work. I will try the channel bank route - just don't
> know too much about the setup options or how you'd purchase the
> correct config, eg. For 150 FXS channel bank, can I get a single PCI
> card for that?
>
> I may end up using the grandstream fxs gateways then use the T1
> channel bank from sangoma,
>
> Thank you all..
>
> Lastly, I know asterisk now has an offical skype_ module, Is there
> anything similar I could use?
>
>
> On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory at voipsupply.com<http://mc/compose?to=cory@voipsupply.com>>
> wrote:
>
> > Samuel - you could go with FXS gateways or channel banks. If you go
> > the gateway route Grandstream or Audiocodes would work fine.
> > Audiocodes are a bit more telco grade. If you have 25 POTS incoming
> > you could use a 24FXO channel bank cross connected with Rhino T1
> > cards, or individual FXO gateways but you may have a hard time
> > finding 24 ports of FXO in a single GW. Best performing T1 cards in
> > my experience (thousands of deployments) are Sangoma. Your server
> > configuration looks fine.
> >
> > Cory J. Andrews
> > Director New Market Initiatives
> >
> > Sayers Media Group
> > VoIP Supply, LLC
> > 454 Sonwil Drive
> > Buffalo, NY 14225
> > 716-250-3402 OFFICE
> > 716-630-1548 FAX
> > 716-601-4474 MOBILE
> > candrews at sayersmedia.com <http://mc/compose?to=candrews@sayersmedia.com>
> >
> >
> > Have I exceeded your expectations? Please share your experience
> > with my boss, Benjamin P. Sayers, CEO
> >
> > NOTICE: The information contained in this email and any document
> > attached hereto is intended only for the named recipient(s). It is
> > the property of the VoIP Supply, LLC and shall not be used,
> > disclosed or reproduced without the express written consent of VoIP
> > Supply, LLC. If you are not the intended recipient, nor the employee
> > or agent responsible for delivering this message in confidence to
> > the intended recipient(s), you are hereby notified that you have
> > received this transmittal in error, and any review, dissemination,
> > distribution or copying of this transmittal or its attachments is
> > strictly prohibited. If you have received this transmittal and/or
> > attachments in error, please notify me immediately by reply e-mail
> > or telephone and then delete this message, including any
> > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
> > 14225 USA.
>
>
>
> >
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces@lists.freeswitch.org>
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces@lists.freeswitch.org>]
> On Behalf Of
> > Samuel Mukoti
> > Sent: Wednesday, November 25, 2009 2:40 PM
> > To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
>
> > Subject: [Freeswitch-users] Grandstream gateways
> >
> > Hi all,
> >
> > I'm wanting to try out a my first large scale setup at the office, 200
> > extensions and 24 POTS incoming, also a T1 line once the telco guys
> > are ready. I wanted assistance with choosing the most appropriate
> > hardware. We already have about 150 analogue phones, and I was
> > wondering what's best? A couple of grandstream FXS GXW4024? Also for
> > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma
> > or digium card? The best voice quality is paramount. Lastly for T1
> > what cards are recommeded,
> >
> > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM,
> > would that perform? Or do I need hardware transcoding?
> >
> > Thank you,
> >
> > Sam
> >
> > Twitter: twitter.com/samuelmukoti
> >
> >
> > On 25 Nov,2009, at 8:05 PM,
> freeswitch-users-request at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-request@lists.freeswitch.org>
> > wrote:
> >
> >> Send FreeSWITCH-users mailing list submissions to
> >> freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >>
> >> To subscribe or unsubscribe via the World Wide Web, visit
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> or, via email, send a message with subject or body 'help' to
> >> freeswitch-users-request at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-request@lists.freeswitch.org>
> >>
> >> You can reach the person managing the list at
> >> freeswitch-users-owner at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-owner@lists.freeswitch.org>
> >>
> >> When replying, please edit your Subject line so it is more specific
> >> than "Re: Contents of FreeSWITCH-users digest..."
> >>
> >>
> >> Today's Topics:
> >>
> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris)
> >> 2. Re: Handling the 302 Moved Temporarily response from
> >> JavaScript (Michael Jerris)
> >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West)
> >> 4. Re: remote_media_ip variable not set (Michael Jerris)
> >> 5. Re: How to find whether the destination extension supports
> >> encryption (Michael Jerris)
> >> 6. Re: Bypass_media and re_invite (srinivasula reddy)
> >> 7. Re: Handling the 302 Moved Temporarily response from
> >> JavaScript (Stephen Crosby)
> >> 8. Re: Handling the 302 Moved Temporarily response from
> >> JavaScript (Tihomir Culjaga)
> >>
> >>
> >> ---
> >> -------------------------------------------------------------------
> >>
> >> Message: 1
> >> Date: Wed, 25 Nov 2009 12:44:46 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike@jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort
> >> invitations
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com<http://mc/compose?to=1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B@jerris.com>
> >
> >> Content-Type: text/plain; charset="windows-1252"
> >>
> >> Its a feature we don't have, patches welcome.
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote:
> >>
> >>> Hi members,
> >>> I?m controlling freeswitch with the conference module via xmlrpc.
> >>>
> >>> Is it desired that the kick command can only kick users that are
> >>> connected to the conference?
> >>> Is there no chance abort an invitation?
> >>> The kick command has no effect until the person I invited with the
> >>> dial command is connected.
> >>
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html
> >>
> >> ------------------------------
> >>
> >> Message: 2
> >> Date: Wed, 25 Nov 2009 12:45:50 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike@jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
> >> response from JavaScript
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID: <A8FA625F-16D2-4A9F-B8C4-13343A488777 at jerris.com<http://mc/compose?to=A8FA625F-16D2-4A9F-B8C4-13343A488777@jerris.com>
> >
> >> Content-Type: text/plain; charset=us-ascii
> >>
> >> In trunk there is a sofia profile setting to allow dialplan
> >> processing of 302 responses. This won't get you back into your same
> >> javascript, but you can probably do something clever from there.
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
> >>
> >>>
> >>> I have considered writing JavaScript code to bridge two calls
> >>> together. However, I would like to perform custom handling of the
> >>> 302 Moved Temporarily response. How do I handle the 302 Moved
> >>> Temporarily response if I use JavaScript?
> >>>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 3
> >> Date: Wed, 25 Nov 2009 11:46:05 -0600
> >> From: Brian West <brian at freeswitch.org<http://mc/compose?to=brian@freeswitch.org>
> >
> >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via
> >> proxy.
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org<http://mc/compose?to=0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org>
> >
> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes
> >>
> >> Yes an alias will be required for every domain you run on the profile
> >> so it can find it.
> >>
> >> /b
> >>
> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
> >>
> >>> Try an alias on the sip profile.
> >>>
> >>> Mike
> >>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 4
> >> Date: Wed, 25 Nov 2009 12:47:37 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike@jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID: <DF3ECA04-0247-40BB-A810-2468F9C4D805 at jerris.com<http://mc/compose?to=DF3ECA04-0247-40BB-A810-2468F9C4D805@jerris.com>
> >
> >> Content-Type: text/plain; charset=us-ascii
> >>
> >> It's possible it does not. I just added some code to set it on auto-
> >> adjust so it might be there sometimes now. You might need to add
> >> some code in mod_sofia to add it other times. Maybe it makes sense
> >> to move that var setting down to switch_rtp.c. Patches for this
> >> would be welcome.
> >>
> >> Thanks
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote:
> >>
> >>> Hi,
> >>>
> >>> In the case of proxy_media=true, does it gets set at all then?
> >>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 5
> >> Date: Wed, 25 Nov 2009 12:48:39 -0500
> >> From: Michael Jerris <mike at jerris.com<http://mc/compose?to=mike@jerris.com>
> >
> >> Subject: Re: [Freeswitch-users] How to find whether the destination
> >> extension supports encryption
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com<http://mc/compose?to=38C9574B-EA25-4B8F-9AF6-21861D0FDA40@jerris.com>
> >
> >> Content-Type: text/plain; charset=us-ascii
> >>
> >> You can send the call with secure enabled and if it supports it it
> >> will use it.
> >>
> >> Mike
> >>
> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote:
> >>
> >>> Hello,
> >>>
> >>> We have a mix of phones that support RTP encryption and those that
> >>> do not. I have to support both types in the meanwhile, and would
> >>> like to have encryption enabled on the relevant leg, even if the
> >>> other leg does not support it (why? one of our ATAs either must
> >>> have it unencrypted or have it encrypted, but cannot have both).
> >>>
> >>> How do I find whether the destination supports encryption? I do not
> >>> want to manage an additional table in the database...
> >>>
> >>
> >>
> >>
> >> ------------------------------
> >>
> >> Message: 6
> >> Date: Wed, 25 Nov 2009 23:25:01 +0530
> >> From: srinivasula reddy <srinivas.ksvreddy at gmail.com<http://mc/compose?to=srinivas.ksvreddy@gmail.com>
> >
> >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID:
> >> <f8af5740911250955x62d66f55h9584582beba76ba0 at mail.gmail.com<http://mc/compose?to=f8af5740911250955x62d66f55h9584582beba76ba0@mail.gmail.com>
> >
> >> Content-Type: text/plain; charset="iso-8859-1"
> >>
> >> HI,
> >> thanks for your reply, my requirement is i am doing failover stuff
> >> with
> >> freeswitch. i dont want cut the calls when freeswitch dies, when
> >> failover
> >> happens mean one freeswitch dies we are going to start the second
> >> freeswitch, i dont want close call intiated by the first
> >> freeswtich, they
> >> are communicating with meida(bypass media). when one endpoing try to
> >> end the
> >> call at that time i want to close the call for the other end also.
> >>
> >>
> >> srinivas
> >>
> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris <mike at jerris.com<http://mc/compose?to=mike@jerris.com>
> >
> >> wrote:
> >>
> >>> FreeSWITCH will kill the calls when you shut it down, if you
> >>> intentionally
> >>> kill the network without shutting down FreeSWITCH the only thing
> >>> you can do
> >>> is enable session timers or rtp timers in the soft phones to kill
> >>> the call
> >>> when FreeSWITCH dies or when the call is over.
> >>>
> >>> Mike
> >>>
> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote:
> >>>
> >>>> Hi All,
> >>>>
> >>>> goodmorning to all, i have a scenario, two pjsua clients are
> >>>> connected
> >>> with Freeswitch and they are in call and bypass_media=true. i
> >>> close the
> >>> Freeswitch server, still they are in call, again i started the
> >>> Freeswitch,
> >>> and registerd these two endpoints, now how can i end the call
> >>> (estabilished
> >>> by the first Freeswitch)? if i call re_invite will it estabilish
> >>> the call
> >>> between two endpoints?
> >>>> any idea?
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users@lists.freeswitch.org>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Srinivasula Reddy K
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html
> >>
> >> ------------------------------
> >>
> >> Message: 7
> >> Date: Wed, 25 Nov 2009 10:01:14 -0800
> >> From: Stephen Crosby <stevecrozz at gmail.com<http://mc/compose?to=stevecrozz@gmail.com>
> >
> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
> >> response from JavaScript
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID:
> >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com<http://mc/compose?to=11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com>
> >
> >> Content-Type: text/plain; charset="utf-8"
> >>
> >> Surprisingly, I've found no way to access the HTTP response status
> >> code
> >> using mod_spidermonkey_curl. I'd love to see this feature added or
> >> discussed
> >> if it already exists and I'm missing it.
> >>
> >> --Stephen
> >>
> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris <mike at jerris.com<http://mc/compose?to=mike@jerris.com>
> >
> >> wrote:
> >>
> >>> In trunk there is a sofia profile setting to allow dialplan
> >>> processing of
> >>> 302 responses. This won't get you back into your same javascript,
> >>> but you
> >>> can probably do something clever from there.
> >>>
> >>> Mike
> >>>
> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
> >>>
> >>>>
> >>>> I have considered writing JavaScript code to bridge two calls
> >>>> together.
> >>> However, I would like to perform custom handling of the 302 Moved
> >>> Temporarily response. How do I handle the 302 Moved Temporarily
> >>> response if
> >>> I use JavaScript?
> >>>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users@lists.freeswitch.org>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> users
> >>> http://www.freeswitch.org
> >>>
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html
> >>
> >> ------------------------------
> >>
> >> Message: 8
> >> Date: Wed, 25 Nov 2009 19:04:56 +0100
> >> From: Tihomir Culjaga <tculjaga at gmail.com<http://mc/compose?to=tculjaga@gmail.com>
> >
> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
> >> response from JavaScript
> >> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users@lists.freeswitch.org>
> >> Message-ID:
> >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com<http://mc/compose?to=65d96fc80911251004l401d5efbl8df3a2ac920207b8@mail.gmail.com>
> >
> >> Content-Type: text/plain; charset="iso-8859-1"
> >>
> >> this is how i do it from the dialplan:
> >>
> >>
> >>
> >>
> >> <extension name="ServiceLookup">
> >> <condition field="destination_number"
> >> expression="^(300030)(.*)|^\+(300030)(.*)">
> >>
> >> <action application="set" data="bPfx=$1$3"/>
> >> <action application="set" data="bNum=$2$4"/>
> >>
> >> <action inline="true" application="set"
> >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/>
> >> <action application="set"
> >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number:
> >> 1:32} :
> >> ${caller_id_number})}"/>
> >>
> >> <action inline="true" application="set"
> >> data="aPfx=${caller_id_number:0:6}"/>
> >> <action inline="true" application="set"
> >> data="aNum=${caller_id_number:6:16}"/>
> >> <action inline="true" application="set"
> >> data="IP_ADDR=${network_addr}:5060"/>
> >>
> >> <action application="lookup_service_destination" data="in $
> >> {aNum},
> >> in $
> >> {aPfx},
> >> in $
> >> {bNum},
> >> in $
> >> {bPfx},
> >> in
> >> ${IP_ADDR},
> >> out
> >> redContact,
> >> out
> >> authResult"/>
> >>
> >> <action application="log" data="INFO ########################
> >> ServiceLookup ########################\n"/>
> >> <action application="log" data="INFO ########################
> >> contact = '${redContact}' ##############\n"/>
> >> <action application="log" data="INFO ########################
> >> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
> >> <action application="log" data="INFO ########################
> >> RADIUS auth = '${authResult}' ##########\n"/>
> >>
> >> <action application="execute_extension" data="doRedirect XML
> >> public"/>
> >> </condition>
> >> </extension>
> >>
> >>
> >> <extension name="doRedirect">
> >> <condition field="destination_number" expression="^doRedirect$"/>
> >> <condition field="${authResult}" expression="^0$|">
> >> <action application="log" data="INFO ########################
> >> RADIUS auth OK!!!' ##########\n"/>
> >> <action application="redirect" data="${red_contact}"/>
> >> <anti-action application="log" data="INFO
> >> ########################
> >> RADIUS auth NOK!! ##########\n"/>
> >> <anti-action application="respond" data="403 Forbidden"/>
> >> </condition>
> >>
> >> </extension>
> >>
> >>
> >>
> >>
> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris <mike at jerris.com<http://mc/compose?to=mike@jerris.com>
> >
> >> wrote:
> >>
> >>> In trunk there is a sofia profile setting to allow dialplan
> >>> processing of
> >>> 302 responses. This won't get you back into your same javascript,
> >>> but you
> >>> can probably do something clever from there.
> >>>
> >>> Mike
> >>>
> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
> >>>
> >>>>
> >>>> I have considered writing JavaScript code to bridge two calls
> >>>> together.
> >>> However, I would like to perform custom handling of the 302 Moved
> >>> Temporarily response. How do I handle the 302 Moved Temporarily
> >>> response if
> >>> I use JavaScript?
> >>>>
> >>>
> >>> _______________________________________________
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> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189
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