Call transfer/forward PSTN-SIP-PSTN
Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2. This is call bridge UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH -> This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH -> 1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN? TIA -Vladimir Rodionov
Hi there, Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re INVITE) and only pass back the media to the network, or pass back signaling also (SIP REFER)? I know several suppliers who support SIP re INVITE but none that support SIP REFER. Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect and http://wiki.freeswitch.org/wiki/Bypass_Media On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov@gmail.com> wrote:
Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2.
This is call bridge
UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN?
TIA
-Vladimir Rodionov
_______________________________________________ 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
Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty sure it is doable because voxeo offers this option for their Voice XML customers but I am not interested in a hosted solution at the time - it is quite expensive. As far as I understood, Voip provider MUST have pstn call transfer feature enabled by telecom provider (AT&T for example) and this should work fine with SIP. -Vladimir Rodionov On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa@gmail.com> wrote:
Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re INVITE) and only pass back the media to the network, or pass back signaling also (SIP REFER)?
I know several suppliers who support SIP re INVITE but none that support SIP REFER.
Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect and http://wiki.freeswitch.org/wiki/Bypass_Media
On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov@gmail.com> wrote:
Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2.
This is call bridge
UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN?
TIA
-Vladimir Rodionov
_______________________________________________ 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
Actually, this is what I need http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN number? -Vladimir Rodionov On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov@gmail.com>wrote:
Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty sure it is doable because voxeo offers this option for their Voice XML customers but I am not interested in a hosted solution at the time - it is quite expensive. As far as I understood, Voip provider MUST have pstn call transfer feature enabled by telecom provider (AT&T for example) and this should work fine with SIP.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa@gmail.com>wrote:
Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re INVITE) and only pass back the media to the network, or pass back signaling also (SIP REFER)?
I know several suppliers who support SIP re INVITE but none that support SIP REFER.
Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect and http://wiki.freeswitch.org/wiki/Bypass_Media
On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov@gmail.com> wrote:
Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2.
This is call bridge
UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN?
TIA
-Vladimir Rodionov
_______________________________________________ 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
Are your calls coming in on TDM or SIP trunks? Are your calls answered by FreeSWITCH before you need to redirect them? On Sat, Aug 8, 2009 at 11:52 AM, Vladimir Rodionov<vladrodionov@gmail.com> wrote:
Actually, this is what I need
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN number?
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov@gmail.com> wrote:
Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty sure it is doable because voxeo offers this option for their Voice XML customers but I am not interested in a hosted solution at the time - it is quite expensive. As far as I understood, Voip provider MUST have pstn call transfer feature enabled by telecom provider (AT&T for example) and this should work fine with SIP.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa@gmail.com> wrote:
Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re INVITE) and only pass back the media to the network, or pass back signaling also (SIP REFER)?
I know several suppliers who support SIP re INVITE but none that support SIP REFER.
Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect and http://wiki.freeswitch.org/wiki/Bypass_Media
On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov@gmail.com> wrote:
Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2.
This is call bridge
UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN?
TIA
-Vladimir Rodionov
_______________________________________________ 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
A call is coming on SIP trunk. From PSTN. I does not need to be answered, actually - I need to do some logic before redirecting call but I can answer call as well It won't break the app logic. -Vladimir Rodionov On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones <pjintheusa@gmail.com> wrote:
Are your calls coming in on TDM or SIP trunks? Are your calls answered by FreeSWITCH before you need to redirect them?
On Sat, Aug 8, 2009 at 11:52 AM, Vladimir Rodionov<vladrodionov@gmail.com> wrote:
Actually, this is what I need
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN number?
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov < vladrodionov@gmail.com> wrote:
Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am
pretty
sure it is doable because voxeo offers this option for their Voice XML customers but I am not interested in a hosted solution at the time - it is quite expensive. As far as I understood, Voip provider MUST have pstn call transfer feature enabled by telecom provider (AT&T for example) and this should work fine with SIP.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa@gmail.com> wrote:
Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re INVITE) and only pass back the media to the network, or pass back signaling also (SIP REFER)?
I know several suppliers who support SIP re INVITE but none that support SIP REFER.
Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect and http://wiki.freeswitch.org/wiki/Bypass_Media
On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<
vladrodionov@gmail.com>
wrote:
Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2.
This is call bridge
UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN?
TIA
-Vladimir Rodionov
_______________________________________________ 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
I don't know of any sip carriers who will let you do refer. you will need to find a carrier who supports it. FreeSWITCH will have no problem sending it but I doubt you will find a carrier who will let you do it easily. Mike On Aug 8, 2009, at 3:12 PM, Vladimir Rodionov wrote:
A call is coming on SIP trunk. From PSTN. I does not need to be answered, actually - I need to do some logic before redirecting call but I can answer call as well It won't break the app logic.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones <pjintheusa@gmail.com> wrote: Are your calls coming in on TDM or SIP trunks? Are your calls answered by FreeSWITCH before you need to redirect them?
On Sat, Aug 8, 2009 at 11:52 AM, Vladimir Rodionov<vladrodionov@gmail.com> wrote:
Actually, this is what I need
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN number?
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov@gmail.com
wrote:
Yes, I need SIP-REFER (pass back media and signaling to PSTN). I
am pretty
sure it is doable because voxeo offers this option for their Voice XML customers but I am not interested in a hosted solution at the time - it is quite expensive. As far as I understood, Voip provider MUST have pstn call transfer feature enabled by telecom provider (AT&T for example) and this should work fine with SIP.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa@gmail.com
wrote:
Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re INVITE) and only pass back the media to the network, or pass back signaling also (SIP REFER)?
I know several suppliers who support SIP re INVITE but none that support SIP REFER.
Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect and http://wiki.freeswitch.org/wiki/Bypass_Media
On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov@gmail.com
wrote:
Good morning, This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes through PSTN Gateway (1) to freeSWITCH application server (AS) (2). AS does some logic and transfers call (or forward) out of Voip provider network to another PSTN number User2.
This is call bridge
UA1 (PSTN) - -> UA2 (PSTN) - - - (1) - (4) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
This is what I want to acomplish (4) UA1 (PSTN) ------------------------------- -> UA2 (PSTN) - - (1) -> PSTN Gateway-> - - (2) - - (3) -> FreeSWITCH ->
1. Can it be implemented in FreeSWITCH? 2. Does anybody know Voip providers which support out of network call transfer/forwarding to PSTN?
TIA
-Vladimir Rodionov
participants (3)
-
Michael Jerris
-
Phillip Jones
-
Vladimir Rodionov