Attended transfer to a conference room
Hi all, I am experiencing a strange issue with SIP based attended transfers. If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS. Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones. I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated. Best, Emrah
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote:
Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Hi MC, Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831 Any idea would be greatly appreciated. Best, Emrah On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
you are missing some of the sip and make sure its GIT HEAD (this should be on jira) sofia global siptrace on On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Oh, And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution. On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
Hi Anthony, The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap. Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Emrah, I think this particular use case should be quite easy to program in a Lua script. The incoming call is handed to the script, the script originates a call to the operator, bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference. on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine. ----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, September 6, 2012 5:57 PM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Hi Anthony,
The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.
Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
-- 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
after a second thought, I realized that you probably want to minimize your own DTMF dialing effort after the call is set up. This can be done easier: you can build additional extensions which only work for your SIP account and connect into the conference rooms without PIN. This can be done (again) with a Lua script, or probably by manipulating the contexts. ----- Original Message -----
From: Stanislav Sinyagin <ssinyagin@yahoo.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Friday, September 7, 2012 12:19 AM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Emrah, I think this particular use case should be quite easy to program in a Lua script. The incoming call is handed to the script, the script originates a call to the operator, bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, September 6, 2012 5:57 PM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Hi Anthony,
The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.
Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
-- 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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Thanks Stanislav, your idea is good, but works only for incoming calls. I know we could trick the idea around to have it work for outbound calls as well, but it sounds to me as being too much of a workaround for something so trivial. I'll play around some more and loop back my calls if I have to. Thanks for this and have a good night, Emrah On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
Emrah, I think this particular use case should be quite easy to program in a Lua script. The incoming call is handed to the script, the script originates a call to the operator, bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, September 6, 2012 5:57 PM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Hi Anthony,
The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.
Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
-- 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Is the issue simply that you need to call into a conference and announce the user before completing the transfer? -MC On Thu, Sep 6, 2012 at 3:39 PM, Emrah <lists@kavun.ch> wrote:
Thanks Stanislav, your idea is good, but works only for incoming calls. I know we could trick the idea around to have it work for outbound calls as well, but it sounds to me as being too much of a workaround for something so trivial.
I'll play around some more and loop back my calls if I have to.
Thanks for this and have a good night, Emrah On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
Emrah, I think this particular use case should be quite easy to program in a Lua script. The incoming call is handed to the script, the script originates a call to the operator, bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, September 6, 2012 5:57 PM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Hi Anthony,
The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.
Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
> Go ahead and clean up the logs and put them on pastebin.freeswitch.org. > -MC > > On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: > Hi all, > > I am experiencing a strange issue with SIP based attended transfers. > > If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. > There is no much activity on the SIP side of things, it seems to be very much related to FS. > > Some info: > I call out through a provider configured on the external profile, from a phone registered on the internal profile. > It is not a codec conflict. > Both lines are answered when I actually finalize the transfer. > I tried with multiple phones and softphones. > > I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated. > > Best, > Emrah >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
-- 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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Yes Sir, it pretty much sums up this use case. I also have phones that only support attended transfers and so it really is my only choice unless I use a PBX feature. On Sep 6, 2012, at 6:44 PM, Michael Collins <msc@freeswitch.org> wrote:
Is the issue simply that you need to call into a conference and announce the user before completing the transfer? -MC
On Thu, Sep 6, 2012 at 3:39 PM, Emrah <lists@kavun.ch> wrote: Thanks Stanislav, your idea is good, but works only for incoming calls. I know we could trick the idea around to have it work for outbound calls as well, but it sounds to me as being too much of a workaround for something so trivial.
I'll play around some more and loop back my calls if I have to.
Thanks for this and have a good night, Emrah On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
Emrah, I think this particular use case should be quite easy to program in a Lua script. The incoming call is handed to the script, the script originates a call to the operator, bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, September 6, 2012 5:57 PM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Hi Anthony,
The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.
Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
> Go ahead and clean up the logs and put them on pastebin.freeswitch.org. > -MC > > On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: > Hi all, > > I am experiencing a strange issue with SIP based attended transfers. > > If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. > There is no much activity on the SIP side of things, it seems to be very much related to FS. > > Some info: > I call out through a provider configured on the external profile, from a phone registered on the internal profile. > It is not a codec conflict. > Both lines are answered when I actually finalize the transfer. > I tried with multiple phones and softphones. > > I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated. > > Best, > Emrah >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
-- 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
for outgoing calls, this could be a different Lua script, or the same script that handles both modes: so, the conference moderator calls the destination number with some unique special prefix, this call gets into the script, then script dials out and connects B leg with the moderator, and after that waits for moderator's input to bridge the called person into a conference. also if I'm not mistaken, I've seen somewhere an extension to the strandard freeswitch conference which allows to dialout and bring new people into the room. in any case, you can easily callout and bring people into the room from FreeSWITCH command line. This can also scripted in some way. ----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Friday, September 7, 2012 12:39 AM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
T hanks Stanislav, your idea is good, but works only for incoming calls. I know we could trick the idea around to have it work for outbound calls as well, but it sounds to me as being too much of a workaround for something so trivial.
I'll play around some more and loop back my calls if I have to.
Thanks for this and have a good night, Emrah On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
Emrah, I think this particular use case should be quite easy to program in a Lua script. The incoming call is handed to the script, the script originates a call to the operator, bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, September 6, 2012 5:57 PM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Hi Anthony,
The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.
Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
> Go ahead and clean up the logs and put them on pastebin.freeswitch.org. > -MC > > On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: > Hi all, > > I am experiencing a strange issue with SIP based attended transfers. > > If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. > There is no much activity on the SIP side of things, it seems to be very much related to FS. > > Some info: > I call out through a provider configured on the external profile, from a phone registered on the internal profile. > It is not a codec conflict. > Both lines are answered when I actually finalize the transfer. > I tried with multiple phones and softphones. > > I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated. > > Best, > Emrah >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- 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
-- 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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 now realize that even blind transfers fail. There must be something wrong in my config. Will post the logs tomorrow. On Sep 6, 2012, at 6:48 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
for outgoing calls, this could be a different Lua script, or the same script that handles both modes:
so, the conference moderator calls the destination number with some unique special prefix, this call gets into the script, then script dials out and connects B leg with the moderator, and after that waits for moderator's input to bridge the called person into a conference.
also if I'm not mistaken, I've seen somewhere an extension to the strandard freeswitch conference which allows to dialout and bring new people into the room.
in any case, you can easily callout and bring people into the room from FreeSWITCH command line. This can also scripted in some way.
----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Friday, September 7, 2012 12:39 AM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
T hanks Stanislav, your idea is good, but works only for incoming calls. I know we could trick the idea around to have it work for outbound calls as well, but it sounds to me as being too much of a workaround for something so trivial.
I'll play around some more and loop back my calls if I have to.
Thanks for this and have a good night, Emrah On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
Emrah, I think this particular use case should be quite easy to program in a Lua script. The incoming call is handed to the script, the script originates a call to the operator, bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.
on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.
----- Original Message -----
From: Emrah <lists@kavun.ch> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, September 6, 2012 5:57 PM Subject: Re: [Freeswitch-users] Attended transfer to a conference room
Hi Anthony,
The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference. Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.
Best, Emrah On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Oh,
And you should try to avoid doing an attended transfer to one-legged-calls like conference or ivr, blind transfers work better for this because calls to apps are not bridged and the concept of transferring becomes confusing. The other alternative is to bridge to the conference by looping the call over loopback or calling to the same box on sip so there is a true bridge. But blind transferring is the best solution.
On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote: > Hi MC, > > Thanks a bunch for your reply, sorry for the delay. > Here are my logs: > http://pastebin.freeswitch.org/19831 > > Any idea would be greatly appreciated. > > Best, > Emrah > > > On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote: > >> Go ahead and clean up the logs and put them on pastebin.freeswitch.org. >> -MC >> >> On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: >> Hi all, >> >> I am experiencing a strange issue with SIP based attended transfers. >> >> If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. >> There is no much activity on the SIP side of things, it seems to be very much related to FS. >> >> Some info: >> I call out through a provider configured on the external profile, from a phone registered on the internal profile. >> It is not a codec conflict. >> Both lines are answered when I actually finalize the transfer. >> I tried with multiple phones and softphones. >> >> I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated. >> >> Best, >> Emrah >>
>> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >>
>> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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
-- 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
-- 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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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, Thank you for taking the time to provide your input. I did not think the referrer's side of SIP was relevant, and so enabled SIPtrace on the external peer only. I will be doing more tests and keep you posted. It feels as if I underwent a complete telephony mindset upgrade ever since I traded Asterisk for FS. Thank you for making it happen. All the best, Emrah On Sep 4, 2012, at 1:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
you are missing some of the sip and make sure its GIT HEAD (this should be on jira)
sofia global siptrace on
On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
Go ahead and clean up the logs and put them on pastebin.freeswitch.org. -MC
On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists@kavun.ch> wrote: Hi all,
I am experiencing a strange issue with SIP based attended transfers.
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS.
Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones.
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.
Best, Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 will be preaching around to have more people pull their heads out of their ASterisk. Love it! On Sep 4, 2012, at 4:04 PM, Michael Collins <msc@freeswitch.org> wrote:
It feels as if I underwent a complete telephony mindset upgrade ever since I traded Asterisk for FS. Thank you for making it happen.
This process is known as pulling your head out of your Asterisk...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
participants (4)
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Anthony Minessale
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Emrah
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Michael Collins
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Stanislav Sinyagin