[Freeswitch-users] Asterisk to FreeSWITCH migration guide

Wed Aug 17 22:00:34 MSD 2011

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
>
> FreeSWITCH http://www.freeswitch.org/
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>
> AIM: anthm
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>
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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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>
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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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