hi there, i would like to ask, if fs has support for something like "SIP Overlap"? instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side? thanks and kind regards dennis
have you tried it? I *think* either we did support it or we would with a small patch to sofia lib that I cannot recall if we ever got committed. On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote:
hi there,
i would like to ask, if fs has support for something like "SIP Overlap"?
instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side?
thanks and kind regards dennis
_______________________________________________ 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not. any hint? 2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>:
have you tried it? I *think* either we did support it or we would with a small patch to sofia lib that I cannot recall if we ever got committed.
On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote:
hi there,
i would like to ask, if fs has support for something like "SIP Overlap"?
instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side?
thanks and kind regards dennis
_______________________________________________ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
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you need a softswitch.... i'm afraid a SIP phone is not designed for overlap... T. On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann@googlemail.com> wrote:
how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not.
any hint?
2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>:
have you tried it? I *think* either we did support it or we would with a small patch to sofia lib that I cannot recall if we ever got committed.
On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote:
hi there,
i would like to ask, if fs has support for something like "SIP Overlap"?
instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side?
thanks and kind regards dennis
_______________________________________________ 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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i do think some softphone can do it but i forgot which one it was either snom or grandstream On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
you need a softswitch.... i'm afraid a SIP phone is not designed for overlap...
T.
On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann@googlemail.com> wrote:
how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not.
any hint?
2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>:
have you tried it? I *think* either we did support it or we would with a small patch to sofia lib that I cannot recall if we ever got committed.
On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote:
hi there,
i would like to ask, if fs has support for something like "SIP
Overlap"?
instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side?
thanks and kind regards dennis
_______________________________________________ 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Both support it. In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing). It can be problematic if you have a device somewhere in the middle that doesn't support 484s. -metik ----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, October 13, 2009 2:01 PM Subject: Re: [Freeswitch-users] SIP Overlap support? i do think some softphone can do it but i forgot which one it was either snom or grandstream On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote: you need a softswitch.... i'm afraid a SIP phone is not designed for overlap... T. On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann@googlemail.com> wrote: how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not. any hint? 2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>: > have you tried it? > I *think* either we did support it or we would with a small patch to sofia > lib that I cannot recall if we ever got committed. > > > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote: >> >> hi there, >> >> i would like to ask, if fs has support for something like "SIP Overlap"? >> >> instead of receiving the phonenumber from our carrier in a block, we >> want to receive the phonenumber digit-by-digit and we want to tell fs >> when the number is complete. our carrier could send us the phonenumber >> digit-by-digit, but what about the fs-side? >> >> >> thanks and kind regards >> dennis >> >> _______________________________________________ >> 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 > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ------------------------------------------------------------------------------ _______________________________________________ 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 never found it working properly... i always had some interoperability issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability. BTW: using overlap can lead to a greater system load... be careful when setting the minimum number of digits you will send in 1st message. I wish you luck... T. On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list@metik.com>wrote:
Both support it. In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing). It can be problematic if you have a device somewhere in the middle that doesn't support 484s.
-metik
----- Original Message ----- *From:* Anthony Minessale <anthony.minessale@gmail.com> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, October 13, 2009 2:01 PM *Subject:* Re: [Freeswitch-users] SIP Overlap support?
i do think some softphone can do it but i forgot which one it was either snom or grandstream
On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
you need a softswitch.... i'm afraid a SIP phone is not designed for overlap...
T.
On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann@googlemail.com> wrote:
how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not.
any hint?
2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>:
have you tried it? I *think* either we did support it or we would with a small patch to sofia lib that I cannot recall if we ever got committed.
On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote:
hi there,
i would like to ask, if fs has support for something like "SIP
Overlap"?
instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side?
thanks and kind regards dennis
_______________________________________________ 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<MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
------------------------------
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As evidenced by various DTMF interop issues (with RFC2833, inband, etc) over the years, I would avoid it if at all possible. What does it particularly do that can not accomplished by using RFC 2833 or (less ideal) inband DTMF? Or are you attempting to use it as a band-aid to address some sort of interop issue with the carrier involved that is wrecking havoc with your particular application? -metik ----- Original Message ----- From: Tihomir Culjaga To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, October 13, 2009 3:24 PM Subject: Re: [Freeswitch-users] SIP Overlap support? i never found it working properly... i always had some interoperability issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability. BTW: using overlap can lead to a greater system load... be careful when setting the minimum number of digits you will send in 1st message. I wish you luck... T. On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list@metik.com> wrote: Both support it. In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing). It can be problematic if you have a device somewhere in the middle that doesn't support 484s. -metik ----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, October 13, 2009 2:01 PM Subject: Re: [Freeswitch-users] SIP Overlap support? i do think some softphone can do it but i forgot which one it was either snom or grandstream On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote: you need a softswitch.... i'm afraid a SIP phone is not designed for overlap... T. On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann@googlemail.com> wrote: how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not. any hint? 2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>: > have you tried it? > I *think* either we did support it or we would with a small patch to sofia > lib that I cannot recall if we ever got committed. > > > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote: >> >> hi there, >> >> i would like to ask, if fs has support for something like "SIP Overlap"? >> >> instead of receiving the phonenumber from our carrier in a block, we >> want to receive the phonenumber digit-by-digit and we want to tell fs >> when the number is complete. our carrier could send us the phonenumber >> digit-by-digit, but what about the fs-side? >> >> >> thanks and kind regards >> dennis >> >> _______________________________________________ >> 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 > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 -------------------------------------------------------------------------- _______________________________________________ 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 suppose he want to have a central dialplan and a dummy phone instead... something as a MGCP phone behavior. T. On Tue, Oct 13, 2009 at 10:22 PM, Metik <freeswitch-users-list@metik.com>wrote:
As evidenced by various DTMF interop issues (with RFC2833, inband, etc) over the years, I would avoid it if at all possible.
What does it particularly do that can not accomplished by using RFC 2833 or (less ideal) inband DTMF? Or are you attempting to use it as a band-aid to address some sort of interop issue with the carrier involved that is wrecking havoc with your particular application?
-metik
----- Original Message ----- *From:* Tihomir Culjaga <tculjaga@gmail.com> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, October 13, 2009 3:24 PM *Subject:* Re: [Freeswitch-users] SIP Overlap support?
i never found it working properly... i always had some interoperability issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability.
BTW: using overlap can lead to a greater system load... be careful when setting the minimum number of digits you will send in 1st message. I wish you luck...
T.
On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-list@metik.com>wrote:
Both support it. In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing). It can be problematic if you have a device somewhere in the middle that doesn't support 484s.
-metik
----- Original Message ----- *From:* Anthony Minessale <anthony.minessale@gmail.com> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, October 13, 2009 2:01 PM *Subject:* Re: [Freeswitch-users] SIP Overlap support?
i do think some softphone can do it but i forgot which one it was either snom or grandstream
On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
you need a softswitch.... i'm afraid a SIP phone is not designed for overlap...
T.
On Tue, Oct 13, 2009 at 5:26 PM, Dennis <odermann@googlemail.com> wrote:
how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not.
any hint?
2009/10/13 Anthony Minessale <anthony.minessale@gmail.com>:
have you tried it? I *think* either we did support it or we would with a small patch to sofia lib that I cannot recall if we ever got committed.
On Tue, Oct 13, 2009 at 8:51 AM, Dennis <odermann@googlemail.com> wrote:
hi there,
i would like to ask, if fs has support for something like "SIP
Overlap"?
instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the
phonenumber
digit-by-digit, but what about the fs-side?
thanks and kind regards dennis
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the thing we want to make working nicer is the following: we want the main/basic phonenumber (123456) to be reachable, so that the telephone rings. but we also want it to be expandable with ddi-digits. example: dial the 123456 to reach the company, dial the 123456 1 to reach the support. in the moment our carrier waits x (milli-)seconds to see, if the 123456 was dialed or if there were more digits attached. after this time, the carrier sends us the whole number as a block (may it be 123456 or 1234561) and we answer. we want to avoid the waiting time the carrier needs to wait for (possible) more digits. we want to receive every single digit from our carrier and we want to tell fs, when the number is complete and fs can answer.
So with overlap you will have to keep refusing the call until the right amount of digits are dialed. This mode would send 1 then 12 then 123 then 1234 then 12345 then 123456 as they were being dialed. once you have 123456 won't you still be unsure if he will type the next 1 or not and be forced to refuse it and wait anyway? This feature is mostly used to allow the server to accept the call the instant it knows there is an exact match in the dialplan or a non-existent number has been dialed. For instance, if 1234 was a unique ext, it could instantly return since it knows. On Wed, Oct 14, 2009 at 9:31 AM, Dennis <odermann@googlemail.com> wrote:
the thing we want to make working nicer is the following:
we want the main/basic phonenumber (123456) to be reachable, so that the telephone rings. but we also want it to be expandable with ddi-digits.
example: dial the 123456 to reach the company, dial the 123456 1 to reach the support.
in the moment our carrier waits x (milli-)seconds to see, if the 123456 was dialed or if there were more digits attached. after this time, the carrier sends us the whole number as a block (may it be 123456 or 1234561) and we answer.
we want to avoid the waiting time the carrier needs to wait for (possible) more digits. we want to receive every single digit from our carrier and we want to tell fs, when the number is complete and fs can answer.
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once you have 123456 won't you still be unsure if he will type the next 1 or not and be forced to refuse it and wait anyway?
basically you are right. BUT, we know, that a basic phone number has 6 digits - so, we do not have to check anything before. as soon as we have 6 digits, we look in our database for this number. a flag in the database tells us, if this number is allowed to use ddi (further digits) or not. if the number is not allowed, we can answer the call directly without any further waiting. other numbers are maked to have 2 ddi's - here we know, that we have to wait for 2 more digits, before we answer. if a number can have 6 digits or 8 digits, you are right, we have to wait anyway. but without having overlap, there is ALWAYS a waiting time. we would like to make the answer-times as short as possible. if no waiting is needed, we do not want to wait. we would like to tell fs to send a 484 response incomplete to the cirpack of our carrier if we need/want more digits. perhaps this works with fs, but not with "socket outbound", which we are using? we know, that this is more work for our servers, but they are powerful enough and have enough resources, that they should be able to handle that. of course we do not want to become to experimental, because fs works so extremly smooth and reliable, that it is nearly unbelievable.
right you can reply 484 in your dp at any time <action application="respond" data="484 Address Incomplete"/> then it should try again. The bit i can't remember is if we committed a certain 1 line patch that makes sofia parse the next invite to the same call properly, the patch was to the sofia lib itself so test it and see. I may need to dig up the answer again from the sofia dev. On Thu, Oct 15, 2009 at 2:47 AM, Dennis <odermann@googlemail.com> wrote:
once you have 123456 won't you still be unsure if he will type the next 1 or not and be forced to refuse it and wait anyway?
basically you are right. BUT, we know, that a basic phone number has 6 digits - so, we do not have to check anything before. as soon as we have 6 digits, we look in our database for this number. a flag in the database tells us, if this number is allowed to use ddi (further digits) or not. if the number is not allowed, we can answer the call directly without any further waiting. other numbers are maked to have 2 ddi's - here we know, that we have to wait for 2 more digits, before we answer. if a number can have 6 digits or 8 digits, you are right, we have to wait anyway.
but without having overlap, there is ALWAYS a waiting time.
we would like to make the answer-times as short as possible. if no waiting is needed, we do not want to wait.
we would like to tell fs to send a 484 response incomplete to the cirpack of our carrier if we need/want more digits. perhaps this works with fs, but not with "socket outbound", which we are using?
we know, that this is more work for our servers, but they are powerful enough and have enough resources, that they should be able to handle that. of course we do not want to become to experimental, because fs works so extremly smooth and reliable, that it is nearly unbelievable.
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ok, we will try this with the cirpack of our carrier. this will take some days, till everything is set up. after the tests i will come back to report. 2009/10/15 Anthony Minessale <anthony.minessale@gmail.com>:
right you can reply 484 in your dp at any time <action application="respond" data="484 Address Incomplete"/>
then it should try again.
The bit i can't remember is if we committed a certain 1 line patch that makes sofia parse the next invite to the same call properly, the patch was to the sofia lib itself so test it and see. I may need to dig up the answer again from the sofia dev.
ok, as written, i come back after some tests with fs and a thomson cirpack. it did not work - at least in our tests. we are using socket outbound and when a call comes in, it starts the socket of fs. the number may be 123456. fs sends the respond 484 and our carrier receives this information. but fs ends the call with hangup_cause = invalid_number_format. the carrier has one more digit for the phone number and sends 1234567 and the above mentioned behavior repeats. the behavior we want and expected is, that the call stays in the socket after response 484, so that the carrier can send the 1234567 into the same socket. the management, when fs should send response 484 and when fs should be answered would be programmed by us. it also important, that fs keeps the call in the socket, so we can tell fs, to answer the call after x seconds anyway. any ideas, what we could do?
This may be possible with tcp, how could this work on udp? Can you provide an rfc reference on this? Mike On Oct 24, 2009, at 8:13 AM, Dennis wrote:
ok, as written, i come back after some tests with fs and a thomson cirpack.
it did not work - at least in our tests.
we are using socket outbound and when a call comes in, it starts the socket of fs. the number may be 123456. fs sends the respond 484 and our carrier receives this information. but fs ends the call with hangup_cause = invalid_number_format. the carrier has one more digit for the phone number and sends 1234567 and the above mentioned behavior repeats.
the behavior we want and expected is, that the call stays in the socket after response 484, so that the carrier can send the 1234567 into the same socket. the management, when fs should send response 484 and when fs should be answered would be programmed by us. it also important, that fs keeps the call in the socket, so we can tell fs, to answer the call after x seconds anyway.
any ideas, what we could do?
On Oct 24, 2009, at 8:13 AM, Dennis wrote:
ok, as written, i come back after some tests with fs and a thomson cirpack.
No idea if this is useful as I'm a noob with fs. If not please excuse the noise. In the past Asterisk to work properly with Cirpack needed the following patch: diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c --- asterisk-1.4.19.org/main/rtp.c 2007-10-08 22:06:33.000000000 +0200 +++ asterisk-1.4.19/main/rtp.c 2007-11-11 13:12:28.000000000 +0100 @@ -1383,6 +1383,7 @@ [34] = {1, AST_FORMAT_H263}, [103] = {1, AST_FORMAT_H263_PLUS}, [97] = {1, AST_FORMAT_ILBC}, + [96] = {0, AST_RTP_DTMF}, [99] = {1, AST_FORMAT_H264}, [101] = {0, AST_RTP_DTMF}, [110] = {1, AST_FORMAT_SPEEX}, Maybe this helps. Regards, Patrick
We have <param name="rfc2833-pt" value="101"/> on the profile and you to set this. So you can set it to 96 if needed. But you shouldn't have to do that if they say 96 and we say 101 they should be listening on 101 from us and we should be listening on 96 from them... thats why its called an RTP map. /b On Oct 31, 2009, at 10:48 AM, Patrick List wrote:
On Oct 24, 2009, at 8:13 AM, Dennis wrote:
ok, as written, i come back after some tests with fs and a thomson cirpack.
No idea if this is useful as I'm a noob with fs. If not please excuse the noise. In the past Asterisk to work properly with Cirpack needed the following patch:
diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c --- asterisk-1.4.19.org/main/rtp.c 2007-10-08 22:06:33.000000000 +0200 +++ asterisk-1.4.19/main/rtp.c 2007-11-11 13:12:28.000000000 +0100 @@ -1383,6 +1383,7 @@ [34] = {1, AST_FORMAT_H263}, [103] = {1, AST_FORMAT_H263_PLUS}, [97] = {1, AST_FORMAT_ILBC}, + [96] = {0, AST_RTP_DTMF}, [99] = {1, AST_FORMAT_H264}, [101] = {0, AST_RTP_DTMF}, [110] = {1, AST_FORMAT_SPEEX},
Maybe this helps.
Regards, Patrick
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hi anthony, i believe, that there is no problem with the communication between fs and the cirpack (everything works to smooth as if this could be possible). if fs sends the 484, the cirpack sends more digits to fs (if there are some), so this works as it should. the problem is, that fs ends the session/socket after a 484, so that the cirpack sends the following digits into another socket. you wrote about a "1 line patch", which might not have been implemented - at least it seems so. is there a way to get someone of the sofia devs to fix this small problem, so that fs sends the 484 without ending the session/socket and waiting for an answer of the cirpack? we would take care of the rest. kind regards, dennis 2009/10/15 Anthony Minessale <anthony.minessale@gmail.com>:
right you can reply 484 in your dp at any time <action application="respond" data="484 Address Incomplete"/>
then it should try again.
The bit i can't remember is if we committed a certain 1 line patch that makes sofia parse the next invite to the same call properly, the patch was to the sofia lib itself so test it and see. I may need to dig up the answer again from the sofia dev.
The patch was it's ability to accept subsequent invites. Your problem is that in sip each new attempt to send an invite is another call. 484 is a final response so the call with too few digits is terminated. On Tue, Nov 3, 2009 at 9:57 AM, Dennis <odermann@googlemail.com> wrote:
hi anthony,
i believe, that there is no problem with the communication between fs and the cirpack (everything works to smooth as if this could be possible). if fs sends the 484, the cirpack sends more digits to fs (if there are some), so this works as it should. the problem is, that fs ends the session/socket after a 484, so that the cirpack sends the following digits into another socket.
you wrote about a "1 line patch", which might not have been implemented - at least it seems so.
is there a way to get someone of the sofia devs to fix this small problem, so that fs sends the 484 without ending the session/socket and waiting for an answer of the cirpack? we would take care of the rest.
kind regards, dennis
2009/10/15 Anthony Minessale <anthony.minessale@gmail.com>:
right you can reply 484 in your dp at any time <action application="respond" data="484 Address Incomplete"/>
then it should try again.
The bit i can't remember is if we committed a certain 1 line patch that makes sofia parse the next invite to the same call properly, the patch was to the sofia lib itself so test it and see. I may need to dig up the answer again from the sofia dev.
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is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. 2009/11/3 Anthony Minessale <anthony.minessale@gmail.com>:
The patch was it's ability to accept subsequent invites. Your problem is that in sip each new attempt to send an invite is another call.
484 is a final response so the call with too few digits is terminated.
I'm going to say No! /b On Nov 4, 2009, at 2:23 AM, Dennis wrote:
is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive?
so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits.
Brian is right, pls, lets stop with exceptions and get stick to RFCs... otherwise it will be a big mess ... T. On Wed, Nov 4, 2009 at 3:03 PM, Brian West <brian@freeswitch.org> wrote:
I'm going to say No!
/b
On Nov 4, 2009, at 2:23 AM, Dennis wrote:
is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive?
so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits.
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You cannot. This is how the sip spec works. Every new invite is a new call and a new trip to the dialplan. You will probably need to design your code to send the appropriate 484 and be prepared to exit and be called again with the new digits. On Wed, Nov 4, 2009 at 2:23 AM, Dennis <odermann@googlemail.com> wrote:
is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive?
so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits.
2009/11/3 Anthony Minessale <anthony.minessale@gmail.com>:
The patch was it's ability to accept subsequent invites. Your problem is that in sip each new attempt to send an invite is another call.
484 is a final response so the call with too few digits is terminated.
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Sorry to dig up an old thread, but ran into this same problem, and trying to understand the RFC as what is actually the expected behaviour. So in the RFC (http://www.ietf.org/rfc/rfc3264.txt) in the Unicast Streams section: For sendrecv RTP
streams, the payload type numbers indicate the value of the payload type field the offerer expects to receive, and would prefer to send
Doesn't that mean if they say 96 and FS says 101, they should be listening on 96 and FS should be listening on 101? Kinda like how ports work, you listen on the port that you offer. Am I completely misinterpreting? Thanks, Dan On Sat, Oct 31, 2009 at 11:58 AM, Brian West <brian@freeswitch.org> wrote:
We have <param name="rfc2833-pt" value="101"/> on the profile and you to set this.
So you can set it to 96 if needed. But you shouldn't have to do that if they say 96 and we say 101 they should be listening on 101 from us and we should be listening on 96 from them... thats why its called an RTP map.
/b
On Oct 31, 2009, at 10:48 AM, Patrick List wrote:
On Oct 24, 2009, at 8:13 AM, Dennis wrote:
ok, as written, i come back after some tests with fs and a thomson cirpack.
No idea if this is useful as I'm a noob with fs. If not please excuse the noise. In the past Asterisk to work properly with Cirpack needed the following patch:
diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c --- asterisk-1.4.19.org/main/rtp.c 2007-10-08 22:06:33.000000000 +0200 +++ asterisk-1.4.19/main/rtp.c 2007-11-11 13:12:28.000000000 +0100 @@ -1383,6 +1383,7 @@ [34] = {1, AST_FORMAT_H263}, [103] = {1, AST_FORMAT_H263_PLUS}, [97] = {1, AST_FORMAT_ILBC}, + [96] = {0, AST_RTP_DTMF}, [99] = {1, AST_FORMAT_H264}, [101] = {0, AST_RTP_DTMF}, [110] = {1, AST_FORMAT_SPEEX},
Maybe this helps.
Regards, Patrick
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Yes thats how I read it... have you downloaded the latest freeswitch? /b On Apr 5, 2010, at 4:54 PM, Dan Le wrote:
Sorry to dig up an old thread, but ran into this same problem, and trying to understand the RFC as what is actually the expected behaviour.
So in the RFC (http://www.ietf.org/rfc/rfc3264.txt) in the Unicast Streams section:
For sendrecv RTP streams, the payload type numbers indicate the value of the payload type field the offerer expects to receive, and would prefer to send
Doesn't that mean if they say 96 and FS says 101, they should be listening on 96 and FS should be listening on 101? Kinda like how ports work, you listen on the port that you offer.
Am I completely misinterpreting?
Thanks, Dan
Oh okay, it's just from your previous comment, it sounded like you said the opposite. I.e., if the other side offered 96, FS would listen on 96. I don't actually have access to the box in question atm, I'm getting this issue second hand. You wouldn't happen to know roughly when this was changed (which you seem to be implying)? The box is definitely running a much older version maybe 1.0.3 or 1.0.4 at the latest. Dan On Mon, Apr 5, 2010 at 5:58 PM, Brian West <brian@freeswitch.org> wrote:
Yes thats how I read it... have you downloaded the latest freeswitch?
/b
On Apr 5, 2010, at 4:54 PM, Dan Le wrote:
Sorry to dig up an old thread, but ran into this same problem, and trying to understand the RFC as what is actually the expected behaviour.
So in the RFC (http://www.ietf.org/rfc/rfc3264.txt) in the Unicast Streams section:
For sendrecv RTP
streams, the payload type numbers indicate the value of the payload type field the offerer expects to receive, and would prefer to send
Doesn't that mean if they say 96 and FS says 101, they should be listening on 96 and FS should be listening on 101? Kinda like how ports work, you listen on the port that you offer.
Am I completely misinterpreting?
Thanks, Dan
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I placed the script in the public context and tried that, but it shows nothing. In the default context, it will show 1002. But in public, it shows nothing. ________________________________ From: David Ponzone <david.ponzone@gmail.com> To: freeswitch-users@lists.freeswitch.org Sent: Mon, April 5, 2010 12:58:03 PM Subject: Re: [Freeswitch-users] how do I get my DID number from lua scriptwith session:getvariable try rdnis But you could put your lua script in your public context, where the calls are handled first. I dont really see the point to do that in default. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
To make sure that the data you are looking for is even available I recommend adding an info app before calling the Lua script: <action application="info"/> Watch the console and it will spit out a bunch of information. Find the data field that contains the information you seek. Check out this page for more information on the different channel variables: http://wiki.freeswitch.org/wiki/Channel_Variables Once you know which channel var has the data you're looking for then it's easy to get it with session:getVariable("var_name") -MC On Tue, Apr 6, 2010 at 12:55 PM, Uzo Uzo <uzoechi@yahoo.com> wrote:
I placed the script in the public context and tried that, but it shows nothing. In the default context, it will show 1002. But in public, it shows nothing.
------------------------------ *From:* David Ponzone <david.ponzone@gmail.com> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, April 5, 2010 12:58:03 PM *Subject:* Re: [Freeswitch-users] how do I get my DID number from lua script with session:getvariable
try rdnis
But you could put your lua script in your public context, where the calls are handled first. I dont really see the point to do that in default.
David Ponzone Direction Technique email: david.ponzone at ipeva.fr <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26www.ipeva.fr - www.ipeva-studio.com
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In the public context, your extension must now look like: <extension name="mylua"> <condition field="destination_number" "expression="^regexp_matching_your_DIDs$"> <action application="lua" data="/usr/local/freeswitch/scripts/ foo.lua"/> </condition> </extension> Inside the script, you should be able to get the DID using the variable destination_number. Another way, simpler I would say, is to do: <extension name="mylua"> <condition field="destination_number" "expression="^(regexp_matching_your_DIDs)$"> <action application="lua" data="/usr/local/freeswitch/scripts/ foo.lua $1"/> </condition> </extension> This way, you get the DID as argv[1] in your LUA script. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 06/04/2010 à 21:55, Uzo Uzo a écrit :
I placed the script in the public context and tried that, but it shows nothing. In the default context, it will show 1002. But in public, it shows nothing.
From: David Ponzone <david.ponzone@gmail.com> To: freeswitch-users@lists.freeswitch.org Sent: Mon, April 5, 2010 12:58:03 PM Subject: Re: [Freeswitch-users] how do I get my DID number from lua script with session:getvariable
try rdnis
But you could put your lua script in your public context, where the calls are handled first. I dont really see the point to do that in default.
David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.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 (11)
-
Anthony Minessale
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Brian West
-
Dan Le
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David Ponzone
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Dennis
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Metik
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Michael Collins
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Michael Jerris
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Patrick List
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Tihomir Culjaga
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Uzo Uzo