[Freeswitch-users] Asterisk to FreeSWITCH migration guide

Thu Aug 18 00:44:47 MSD 2011

Sam,

Did you already pastebin a copy of your script and dialplan? I know we had
talked about it. In any case, I'm hoping to see what you're doing so that we
can offer you some alternative ideas.

-MC

On Wed, Aug 17, 2011 at 11:53 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I don't see much difference in terms of "originate_disposition" when
> calling answer explicitly opposed to not calling it, so since it appears
> there is more issues not calling it I  guess for now I should just call it.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 17, 2011 11:00 AM
>
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> This is another problem related to the callflow of the provider that can be
> fixed.
>
> In an ideal world, using the defaults, when the early media comes up on the
> b leg it will pass to the a leg which also will start sending early media
> and it will happily pass through.
>
> My hunch is they have calls to you set on some kine of LCR hunt that is
> misconfigured and it's trying to get the answer to stop hunting which is not
> right.
>
>
> On Wed, Aug 17, 2011 at 12:32 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I have also found a side effect when I do not explicitly call answer on the
> inbound leg for b-leg calls that do not return "answer" when using another
> DID provider (VOIPInnovations). The side effect is that the a-leg can hear
> the telco network messages from the carrier like "I'm sorry the number you
> dialed is not a working number ..." or "The user is not accepting calls at
> the moment."
>
> If I do explicitly call answer, then I cannot hear those telco messages,
> which would seem to be better fitting for my case.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 17, 2011 9:16 AM
>
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
> On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> The DID provider I am using is from iCall, and I was searching through
> their website and noticed that they mentioned a quote with your name on it
> http://carriers.icall.com/open-source/
> so it appears you have had experience with them.
>
> We have a lot of experience with iCall. I'm not familiar with any hard
> requirement to "answer" the inbound leg prior to bridging an outbound leg.
> What happens in your dialplan if you don't explicitly answer the inbound leg
> prior to calling the bridge app?
> -MC
>
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 16, 2011 5:29 PM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> You should never answer a call before bridging it anyway, it breaks all of
> the accounting.
> It would make sense to find out why the provider is doing that and get it
> fixed.
>
>
> On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
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> --
> Anthony Minessale II
>
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> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> googletalk:conf+888 at conference.freeswitch.org
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> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
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