[Freeswitch-users] Asterisk to FreeSWITCH migration guide

Discussion:

Nestor A Diaz

2011-08-09 16:48:00 UTC

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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Avi Marcus

2011-08-09 23:11:12 UTC

Permalink

A lot of this information is on the freeswitch wiki here:
http://wiki.freeswitch.org/wiki/Rosetta_stone
CDR management: http://wiki.freeswitch.org/wiki/Cdr
Web GUIs:
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_Is_there_a_GUI_for_configuring_FreeSWITCH.3F
Queues: mod_fifo or mod_callcenter

-Avi Marcus

Post by Nestor A Diaz
**
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
*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
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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Nestor A Diaz

2011-08-10 00:19:53 UTC

Permalink

Talking about GUIs, which one do you recommend ?

* blue.box
* fusionpbx
* wikipbx

I haven't found any screenshos of blue.box, does anybody know where i
can take a look at them ? wikipbx seems dead, fusionpbx seems to be
enought for my requirements, but what about blue.box ?

Thanks.

Post by Avi Marcus
http://wiki.freeswitch.org/wiki/Rosetta_stone
CDR management: http://wiki.freeswitch.org/wiki/Cdr
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_Is_there_a_GUI_for_configuring_FreeSWITCH.3F
Queues: mod_fifo or mod_callcenter

[...]
--
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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Sam

2011-08-10 00:54:09 UTC

Permalink

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

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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Nandy Dagondon

2011-08-10 02:07:47 UTC

Permalink

hi nestor,

you'll find more information here:
http://wiki.freeswitch.org/wiki/Specsheet

for Web management: fusionpbx, bluebox

-nandy

Post by Sam
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
*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
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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Avi Marcus

2011-08-10 08:15:24 UTC

Permalink

Easier to do what? fs_cli -x "sofia status" or "show channels" or whatever
and you can do "as xml" so you can parse it easier.
But better is to just pick up a library for your language to make the ESL
stuff much easier.

-Avi

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Moe Navid

2011-08-10 08:44:18 UTC

Permalink

There is no way by any means to compare Asterisk's AGI with the different facilities FreeSWITCH offers you in terms of controlling your call flow.

For almost 3 years I managed a cluster of Asterisk + AGI + AMI with tones of channel locks and core dumps? Asterisk's dial status might seem compelling when you want to do simple things like calling cards etc? but when it comes to complex accounting and routing sky is limitless with the power of FreeSWITCH.

I found FreeSWITCH's learning curve to be like vim, initially it may seem a bit difficult but in long run it pays of very well.

If you know the difference between Dial command in Asterisk and Bridge in FreeSWITCH you would never go back to Asterisk. I give you just 3 simple examples:
1) Bridge command (via the channel variables) gives you the ability to control PDD on calls. Asterisk does not have such facility nonetheless it does not even bother to give you any useful information about your "Dial Status"! To control the PDD I had to tweak my kamailio.

2) If you want to implement a simple rate engine + fail over routing with asterisk + agi for failover you have to have a loop and watch for CONGESTIONs to select your next route/carrier where as in FreeSWITCH you can just simply define your fail overs in your bridge args.

3) If you are in a cluster, have multiple gateways acting as proxy and you want to define outbound proxy for your carriers/endpoints you either have to define bunch of sip peers with outbound proxies or do it in dirty way which I did, I used to add a header in my outgoing calls X-Carrier-IP and had my kamailio to take care of the rest. In FreeSWITCH you just simply add ;fspath= to your bridge args.

List can go on and on and on?

Asterisk's dial status was the most annoying part of asterisk in my opinion :)

Post by Sam
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.
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
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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Sam

2011-08-10 16:12:56 UTC

Permalink

So have you had to retrieve the dial status from bridging a call in freeswitch? For the life of me I cannot properly get the answered_time? when looking up the channel variables after the bridge call finishes an answered call.

________________________________
From: Moe Navid <manavid at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 10, 2011 1:44 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide

There is no way by any means to compare Asterisk's AGI with the different facilities FreeSWITCH offers you in terms of controlling your call flow.

For almost 3 years I managed a cluster of Asterisk + AGI + AMI with tones of channel locks and core dumps? Asterisk's dial status might seem compelling when you want to do simple things like calling cards etc? but when it comes to complex accounting and routing sky is limitless with the power of FreeSWITCH.

I found FreeSWITCH's learning curve to be like vim, initially it may seem a bit difficult but in long run it pays of very well. ?

If you know the difference between Dial command in Asterisk and Bridge in FreeSWITCH you would never go back to Asterisk. I give you just 3 simple examples:
1) Bridge command (via the channel variables) gives you the ability to control PDD on calls. Asterisk does not have such facility nonetheless it does not even bother to give you any useful information about your "Dial Status"! To control the PDD I had to tweak my kamailio.

2) If you want to implement a simple rate engine + fail over routing with asterisk + agi for failover you have to have a loop and watch for CONGESTIONs to select your next route/carrier where as in FreeSWITCH you can just simply define your fail overs in your bridge args.

3) If you are in a cluster, have multiple gateways acting as proxy and you want to define outbound proxy for your carriers/endpoints you either have to define bunch of sip peers with outbound proxies or do it in dirty way which I did, I used to add a header in my outgoing calls X-Carrier-IP and had my kamailio to take care of the rest. In FreeSWITCH you just simply add ;fspath= to your bridge args.

List can go on and on and on?

Asterisk's dial status was the most annoying part of asterisk in my opinion?:)

On Aug 9, 2011, at 5:54 PM, Sam 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.

Post by Nestor A Diaz
________________________________
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.

Post by Nestor A Diaz
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.

Post by Nestor A Diaz
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.

Post by Nestor A Diaz
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.

Post by Nestor A Diaz
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.

Post by Nestor A Diaz
So could you please freeswitch users help me with this rosetta

stone migration guide in order to post it to voip-info.org or

Post by Nestor A Diaz
Technology Asterisk Freeswitch
PSTN Connectivity (Digium /

Sangoma) dahdi freetdm

Post by Nestor A Diaz
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
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Anthony Minessale

2011-08-11 19:55:34 UTC

Permalink

in your cdr records, all of the variables are present there and they are
template-able
freeswitch separates the logic from the call handling.

Post by Sam
So have you had to retrieve the dial status from bridging a call in
freeswitch? For the life of me I cannot properly get the answered_time when
looking up the channel variables after the bridge call finishes an answered
call.
------------------------------
*From:* Moe Navid <manavid at gmail.com>
*To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Sent:* Wednesday, August 10, 2011 1:44 AM
*Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
There is no way by any means to compare Asterisk's AGI with the different
facilities FreeSWITCH offers you in terms of controlling your call flow.
For almost 3 years I managed a cluster of Asterisk + AGI + AMI with tones
of channel locks and core dumps? Asterisk's dial status might seem
compelling when you want to do simple things like calling cards etc? but
when it comes to complex accounting and routing sky is limitless with the
power of FreeSWITCH.
I found FreeSWITCH's learning curve to be like vim, initially it may seem a
bit difficult but in long run it pays of very well.
If you know the difference between Dial command in Asterisk and Bridge in
FreeSWITCH you would never go back to Asterisk. I give you just 3 simple
1) Bridge command (via the channel variables) gives you the ability to
control PDD on calls. Asterisk does not have such facility nonetheless it
does not even bother to give you any useful information about your "Dial
Status"! To control the PDD I had to tweak my kamailio.
2) If you want to implement a simple rate engine + fail over routing with
asterisk + agi for failover you have to have a loop and watch for
CONGESTIONs to select your next route/carrier where as in FreeSWITCH you can
just simply define your fail overs in your bridge args.
3) If you are in a cluster, have multiple gateways acting as proxy and you
want to define outbound proxy for your carriers/endpoints you either have to
define bunch of sip peers with outbound proxies or do it in dirty way which
I did, I used to add a header in my outgoing calls X-Carrier-IP and had my
kamailio to take care of the rest. In FreeSWITCH you just simply add
;fspath= to your bridge args.
List can go on and on and on?
Asterisk's dial status was the most annoying part of asterisk in my opinion :)
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
*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
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Anthony Minessale

2011-08-10 15:52:43 UTC

Permalink

=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........

Sam

2011-08-10 16:28:39 UTC

Permalink

Thanks for being so accommodating. I was a bit frustrated in trying to port over an asterisk agi script to freeswitch. I have spent many hours trying to learn how to configure freeswitch, I was about to give up, but I will play with the new changes you made and see if that works for me.

One other question, when the bridged call hangs up I do not see any value for the hangup time when using getVariable("hangup_time"), so how can I get it?

________________________________
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.

Post by Nestor A Diaz
Fixing those problems either by applying patches or by changing

Post by Nestor A Diaz
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.

Post by Nestor A Diaz
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.

Post by Nestor A Diaz
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.

Post by Nestor A Diaz
So could you please freeswitch users help me with this rosetta

stone migration guide in order to post it to voip-info.org or

Post by Nestor A Diaz
Technology Asterisk Freeswitch
PSTN Connectivity (Digium /

Sangoma) dahdi freetdm

--
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

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-------------- next part --------------
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Nandy Dagondon

2011-08-11 02:40:31 UTC

Permalink

hi sam,

i found this post
http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8477

modify the script to suit your need.

hope it helps. just dig on w/ FS :-)
-nandy

Post by Sam
Thanks for being so accommodating. I was a bit frustrated in trying to port
over an asterisk agi script to freeswitch. I have spent many hours trying to
learn how to configure freeswitch, I was about to give up, but I will play
with the new changes you made and see if that works for me.
One other question, when the bridged call hangs up I do not see any value
for the hangup time when using getVariable("hangup_time"), so how can I get
it?
------------------------------
*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........
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
*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
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
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
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Sam

2011-08-11 17:52:07 UTC

Permalink

Michael,

Were you referring to the link http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html? because the page no longer exists.

________________________________
From: Nandy Dagondon <gcd at i.ph>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 10, 2011 7:40 PM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide

hi sam,

i found this posthttp://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8477

modify the script to suit your need.

hope it helps. ?just dig on w/ FS :-)
-nandy

On Thu, Aug 11, 2011 at 12:28 AM, Sam <lakersman2006 at yahoo.com> wrote:

Thanks for being so accommodating. I was a bit frustrated in trying to port over an asterisk agi script to freeswitch. I have spent many hours trying to learn how to configure freeswitch, I was about to give up, but I will play with the new changes you made and see if that works for me.

Post by Sam
One other question, when the bridged call hangs up I do not see any value for the hangup time when using getVariable("hangup_time"), so how can I get it?
________________________________
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........
?
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.