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. On 08/09/2011 06:11 PM, Avi Marcus wrote:

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_configu... 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@tiendalinux.com Email/MSN: nestor@tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

hi nestor, you'll find more information here: http://wiki.freeswitch.org/wiki/Specsheet for Web management: fusionpbx, bluebox -nandy On Wed, Aug 10, 2011 at 8:54 AM, Sam <lakersman2006@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@tiendalinux.com> *To:* freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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.

From: Nestor A Diaz <nestor@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

in your cdr records, all of the variables are present there and they are template-able freeswitch separates the logic from the call handling. On Wed, Aug 10, 2011 at 11:12 AM, Sam <lakersman2006@yahoo.com> wrote:

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@gmail.com> *To:* FreeSWITCH Users Help <freeswitch-users@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.

------------------------------ *From:* Nestor A Diaz <nestor@tiendalinux.com> *To:* freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

Michael, Were you referring to the link http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-hea...  because the page no longer exists. ________________________________ From: Nandy Dagondon <gcd@i.ph> To: FreeSWITCH Users Help <freeswitch-users@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@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.

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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

On Mon, Aug 15, 2011 at 3:17 PM, Sam <lakersman2006@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.

I think we understand where you are coming from. I believe in my other post that I mentioned that the bridge app and a few other channel variables will most likely let you tailor your dialplan to your exact needs without explicitly needing to poll the "dial status" of the b-leg. In fact, I think I left out a few options: execute_on_answer execute_on_ring execute_on_media Those channel variables are quite handy, especially the execute_on_answer. If you execute a dp transfer when the b leg is answered then whatever happens after your bridge (or originate API if you are doing that) will always be some sort of failed call attempt. There's also a handy "transfer_on_fail" channel variable that lets you explicitly send the call to another dp extension on a failed bridge attempt. It may seem unintuitive to be transferring the calls all over the dialplan, but if you think about it you can create contexts to handle specific scenarios and then you're done. Call failed? Transfer to "CALL_FAILED" context and process. Call was answered? Transfer to "CALL_ANSWERED" context and process. No scripting required, and it's really fast. Just my $0.02... -MC

------------------------------ *From:* Anthony Minessale <anthony.minessale@gmail.com> *To:* FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> *To:* freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

FreeSWITCH-users mailing list 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

How come in some of the examples I see it calling answer()? http://wiki.freeswitch.org/wiki/Perl_Console_IVR_Example ________________________________ From: Anthony Minessale <anthony.minessale@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

FreeSWITCH-users mailing list 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

-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 FreeSWITCH-users mailing list 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

My app is basically a calling card app which is kind of like that DISA example, correct? ________________________________ From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, August 17, 2011 9:14 AM Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide On Tue, Aug 16, 2011 at 6:19 PM, Sam <lakersman2006@yahoo.com> wrote: How come in some of the examples I see it calling answer()?

http://wiki.freeswitch.org/wiki/Perl_Console_IVR_Example

The above example is a DISA-like function. The *only* way it would work is for FreeSWITCH to answer the call. It's an IVR, therefore there is no b-leg. -MC  

________________________________ From: Anthony Minessale <anthony.minessale@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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

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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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006@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@gmail.com> *To:* FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> *To:* FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> *To:* freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

FreeSWITCH-users mailing list 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

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

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@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

FreeSWITCH-users mailing list 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

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@freeswitch.org>

To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

FreeSWITCH-users mailing list 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

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 FreeSWITCH-users mailing list 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

MC, Here is the pastebin http://pastebin.freeswitch.org/17069 of my perl script. For the current script I just made it display the various channel variables so I can see what values freeswitch will provide me after the call has been bridged or not. But the script will be used for a call card app that will just bridge 1 caller to 1 callee, pretty straight forward. So I will need to be able to play back to the caller on certain call states like NO_ANSWER, BUSY, CONGESTION, ETC. ________________________________ From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, August 17, 2011 1:44 PM Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide 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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@freeswitch.org>

To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

FreeSWITCH-users mailing list 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

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

FreeSWITCH-users mailing list 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

FreeSWITCH-users mailing list 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

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Okay, wonderful, thanks for the update. ________________________________ From: Anthony Minessale <anthony.minessale@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, August 17, 2011 9:34 AM Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide I asked iCall and they have acknowledged your issue and there is someone looking into it. On Wed, Aug 17, 2011 at 12:14 AM, Sam <lakersman2006@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.

________________________________ From: Anthony Minessale <anthony.minessale@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@tiendalinux.com> To: freeswitch-users@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@tiendalinux.com Email/MSN: nestor@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@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@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

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On 08/10/2011 04:51 AM, Steven Ayre wrote:

As with Asterisk there's no official GUI, but there are several open source projects providing one. Or you can build your own. A few include blue.box (2800hz project) and FusionPBX. http://wiki.freeswitch.org/wiki/Freeswitch_Gui

Hi, i know there is always the possibility to build my own, but that's not the point, i prefer something ready to use, i neither use those kind of GUIs, those are for my customers, as i always prefer working on a command line and i don't have plans to change my mind regarding this.

As for G729 which you mentioned... do NOT use the Intel IPP codec. It is ILLEGAL unless you have purchased a valid licence for it, which is extremely unlikely. You can support it using a hardware transcoding card (Sangoma), mod_com_g729 (version licensed by FreeSWITCH) and mod_g729 (which is passthrough only, no transcoding but fine for bridging calls).

I know the use of the free g.729 is illegal, but if you want to test it without going to production i don't see the reason why i can't use it on my batcave.

chan_mobile's closest match is probably mod_gsmopen. I believe it uses a cable rather than bluetooth though, and is faily new so probably 'experimental'.

Comming from the asterisk world i understand experimental is the same as not working, beta = maybe, and first stable means some release behind for production environments. Anyway chan_mobile is a hack module but works fine for one or two cell phones. Anybody have used mod_gsmopen ? does this thing really works ? do you have to transfer the dial number tones for making a call ? it that's true i still prefer a sip gateway and a cell plant. I would love to have something like asterisk's chan_sebi running under freeswitch (it never works for me on asterisk but the idea is nice, very nice)

IAX2 is supported by mod_opal. As you noted though, it isn't as stable as say mod_sofia.

discarded, i prefer to use asterisk in another machine for that and brigde the calls to asterisk via SIP. (see chan_mobiles comments for explanation :) )

For fax handling check the T38 functionality provided by mod_spandsp.

T.38 seems good, anybody have been able to make it work with hylafax ?? is possible ? Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor@tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

¿Why use Digium, when Sangoma Cards are fully compatible with FreeSwitch?

That really depends on the app. If the module providing the app hasn't been loaded you'll get an error in your log and it'll skip to the next app. For apps that are loaded, they'll generally either do something or set channel variables that indicate the result. Check the wiki documentation on http://wiki.freeswitch.org/ -Steve On 11 August 2011 07:59, king2kin <xing2kin@yahoo.com> wrote:

Hi folks,

I looked into a few modules (e.g. mod_db, mod_hash, mod_unimrcp, mod_skel and mod_say_xx), and find out that unlike module api, a module app seems not able to return any data. If so, how do we know operation result after we submit app request to FS? for example,

<action application="appx" data="run/my/data"/>

how do I know the above app task is executed successfully or not?

xk

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