On 10 September 2010 08:06, Moises Silva <moises.silva@gmail.com> wrote:
on-hook/off-hook done quickly.
(changed the subject) The that is correct. Often it is called "Recall" or "PBX Recall". It is required if you want to transfer calls etc from analogue phones without the PBX listening for DTMF tones inband. On Asterisk, the idea is if you push this button, you can the dial tone, and can make a new call while the old call is on hold. If you push it again you get a conference call, and then if you hang up you have effectively made an attended transfer of the call.
Brian, what is it that does not work with flash?
When I push the flash button, I get loss of audio for a moment, and then the current call resumes. Admittedly I am not entirely sure what to expect with Freeswitch, but would be nice if I could use it to transfer calls. Same as with Asterisk. Although I never could get this working with Asterisk either, I ended up having to push the off-hook button manually, which is risky (push it for too long and you disconnect the call) and not always possible with modern phones. Changing the time intervals never seemed to help. -- Brian May <brian@microcomaustralia.com.au>
see the default config for analog spans see the enable-analog-option param? choose one of the 2 call-swap or 3-way I think you need to have at least one enabled for flash to do anything in either case you should be able to hit flash, dial a number, hangup for blind xfers call-swap lets make 2 calls and go back and forth 3-way lets you make a 2nd call and flash again to combine them. are you using openzap or freetdm it should be the same in both..... <analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans> On Thu, Sep 9, 2010 at 6:06 PM, Brian May <brian@microcomaustralia.com.au> wrote:
On 10 September 2010 08:06, Moises Silva <moises.silva@gmail.com> wrote:
on-hook/off-hook done quickly.
(changed the subject)
The that is correct. Often it is called "Recall" or "PBX Recall". It is required if you want to transfer calls etc from analogue phones without the PBX listening for DTMF tones inband.
On Asterisk, the idea is if you push this button, you can the dial tone, and can make a new call while the old call is on hold. If you push it again you get a conference call, and then if you hang up you have effectively made an attended transfer of the call.
Brian, what is it that does not work with flash?
When I push the flash button, I get loss of audio for a moment, and then the current call resumes.
Admittedly I am not entirely sure what to expect with Freeswitch, but would be nice if I could use it to transfer calls.
Same as with Asterisk. Although I never could get this working with Asterisk either, I ended up having to push the off-hook button manually, which is risky (push it for too long and you disconnect the call) and not always possible with modern phones. Changing the time intervals never seemed to help. -- Brian May <brian@microcomaustralia.com.au>
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I have the 3way and callswap outside the analog spans, so I would assume it would apply to both? I do think I can flash the hook and make another call, but not be able to answer the call waiting with analog spans. Anthony Minessale <anthony.minessale@gmail.com> wrote:
see the default config for analog spans
see the enable-analog-option param? choose one of the 2 call-swap or 3-way
I think you need to have at least one enabled for flash to do anything
in either case you should be able to hit flash, dial a number, hangup for blind xfers call-swap lets make 2 calls and go back and forth 3-way lets you make a 2nd call and flash again to combine them.
are you using openzap or freetdm it should be the same in both.....
<analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans>
On Thu, Sep 9, 2010 at 6:06 PM, Brian May <brian@microcomaustralia.com.au> wrote:
On 10 September 2010 08:06, Moises Silva <moises.silva@gmail.com> wrote:
on-hook/off-hook done quickly.
(changed the subject)
The that is correct. Often it is called "Recall" or "PBX Recall". It is required if you want to transfer calls etc from analogue phones without the PBX listening for DTMF tones inband.
On Asterisk, the idea is if you push this button, you can the dial tone, and can make a new call while the old call is on hold. If you push it again you get a conference call, and then if you hang up you have effectively made an attended transfer of the call.
Brian, what is it that does not work with flash?
When I push the flash button, I get loss of audio for a moment, and then the current call resumes.
Admittedly I am not entirely sure what to expect with Freeswitch, but would be nice if I could use it to transfer calls.
Same as with Asterisk. Although I never could get this working with Asterisk either, I ended up having to push the off-hook button manually, which is risky (push it for too long and you disconnect the call) and not always possible with modern phones. Changing the time intervals never seemed to help. -- Brian May <brian@microcomaustralia.com.au>
_______________________________________________ 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
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
no its only valid inside the span, it must be set per span. we don't have it implemented to inherit from the higher tag. On Thu, Sep 9, 2010 at 7:14 PM, <covici@ccs.covici.com> wrote:
I have the 3way and callswap outside the analog spans, so I would assume it would apply to both? I do think I can flash the hook and make another call, but not be able to answer the call waiting with analog spans.
Anthony Minessale <anthony.minessale@gmail.com> wrote:
see the default config for analog spans
see the enable-analog-option param? choose one of the 2 call-swap or 3-way
I think you need to have at least one enabled for flash to do anything
in either case you should be able to hit flash, dial a number, hangup for blind xfers call-swap lets make 2 calls and go back and forth 3-way lets you make a 2nd call and flash again to combine them.
are you using openzap or freetdm it should be the same in both.....
<analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans>
On Thu, Sep 9, 2010 at 6:06 PM, Brian May <brian@microcomaustralia.com.au> wrote:
On 10 September 2010 08:06, Moises Silva <moises.silva@gmail.com> wrote:
on-hook/off-hook done quickly.
(changed the subject)
The that is correct. Often it is called "Recall" or "PBX Recall". It is required if you want to transfer calls etc from analogue phones without the PBX listening for DTMF tones inband.
On Asterisk, the idea is if you push this button, you can the dial tone, and can make a new call while the old call is on hold. If you push it again you get a conference call, and then if you hang up you have effectively made an attended transfer of the call.
Brian, what is it that does not work with flash?
When I push the flash button, I get loss of audio for a moment, and then the current call resumes.
Admittedly I am not entirely sure what to expect with Freeswitch, but would be nice if I could use it to transfer calls.
Same as with Asterisk. Although I never could get this working with Asterisk either, I ended up having to push the off-hook button manually, which is risky (push it for too long and you disconnect the call) and not always possible with modern phones. Changing the time intervals never seemed to help. -- Brian May <brian@microcomaustralia.com.au>
_______________________________________________ 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
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
The parameters are in settings, but I put them inside the spans anyway. No change that I could notice. Here is what I want to do. I make a call using the fxo line to an outside number. Someone calls me which rings through the same fxo module. I hear the call waiting beep on the phone. I flash the hook and get a dialtone. What magic do I have to do to flash the fxo channel so I can answer the call. In asterisk they have hard coded *0 to do this, but how do I do the equivalent in fs to flash the fxo from my analog phone connected to the fxs module or from anywhere else for that matter. Anthony Minessale <anthony.minessale@gmail.com> wrote:
no its only valid inside the span, it must be set per span. we don't have it implemented to inherit from the higher tag.
On Thu, Sep 9, 2010 at 7:14 PM, <covici@ccs.covici.com> wrote:
I have the 3way and callswap outside the analog spans, so I would assume it would apply to both? I do think I can flash the hook and make another call, but not be able to answer the call waiting with analog spans.
Anthony Minessale <anthony.minessale@gmail.com> wrote:
see the default config for analog spans
see the enable-analog-option param? choose one of the 2 call-swap or 3-way
I think you need to have at least one enabled for flash to do anything
in either case you should be able to hit flash, dial a number, hangup for blind xfers call-swap lets make 2 calls and go back and forth 3-way lets you make a 2nd call and flash again to combine them.
are you using openzap or freetdm it should be the same in both.....
<analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans>
On Thu, Sep 9, 2010 at 6:06 PM, Brian May <brian@microcomaustralia.com.au> wrote:
On 10 September 2010 08:06, Moises Silva <moises.silva@gmail.com> wrote:
on-hook/off-hook done quickly.
(changed the subject)
The that is correct. Often it is called "Recall" or "PBX Recall". It is required if you want to transfer calls etc from analogue phones without the PBX listening for DTMF tones inband.
On Asterisk, the idea is if you push this button, you can the dial tone, and can make a new call while the old call is on hold. If you push it again you get a conference call, and then if you hang up you have effectively made an attended transfer of the call.
Brian, what is it that does not work with flash?
When I push the flash button, I get loss of audio for a moment, and then the current call resumes.
Admittedly I am not entirely sure what to expect with Freeswitch, but would be nice if I could use it to transfer calls.
Same as with Asterisk. Although I never could get this working with Asterisk either, I ended up having to push the off-hook button manually, which is risky (push it for too long and you disconnect the call) and not always possible with modern phones. Changing the time intervals never seemed to help. -- Brian May <brian@microcomaustralia.com.au>
_______________________________________________ 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
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.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
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
On 10 September 2010 10:30, Anthony Minessale <anthony.minessale@gmail.com> wrote:
no its only valid inside the span, it must be set per span. we don't have it implemented to inherit from the higher tag.
In this case, the sample config is wrong. <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> </settings> ... spans omitted here... </configuration> -- Brian May <brian@microcomaustralia.com.au>
I've gone insane. I checked the code and I did actually make those params work both globally and per-span. On Thu, Sep 9, 2010 at 10:14 PM, Brian May <brian@microcomaustralia.com.au> wrote:
On 10 September 2010 10:30, Anthony Minessale <anthony.minessale@gmail.com> wrote:
no its only valid inside the span, it must be set per span. we don't have it implemented to inherit from the higher tag.
In this case, the sample config is wrong.
<configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> </settings> ... spans omitted here... </configuration>
-- Brian May <brian@microcomaustralia.com.au>
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
On 11 September 2010 02:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I've gone insane. I checked the code and I did actually make those params work both globally and per-span.
Something not right with my setup, when I push flash I get the following messages: 2010-09-11 15:21:22.846530 [WARNING] ftmod_zt.c:1051 Unhandled event 14 for 2:1 2010-09-11 15:21:22.846530 [DEBUG] ftmod_analog.c:864 [s2c1][1:2] Received event [INVALID] in state [UP] 2010-09-11 15:21:22.846530 [DEBUG] ftmod_analog.c:952 [s2c1][1:2] Ignoring event [INVALID] in state [UP] 2010-09-11 15:21:23.095496 [WARNING] ftmod_zt.c:1051 Unhandled event 65585 for 2:1 2010-09-11 15:21:23.095496 [DEBUG] ftmod_analog.c:864 [s2c1][1:2] Received event [INVALID] in state [UP] 2010-09-11 15:21:23.095496 [DEBUG] ftmod_analog.c:952 [s2c1][1:2] Ignoring event [INVALID] in state [UP] which seems to imply it is detecting the even correctly (?), but doesn't know what to do with it, despite having enable-analog-option set. -- Brian May <brian@microcomaustralia.com.au>
On 11 September 2010 15:26, Brian May <brian@microcomaustralia.com.au> wrote:
which seems to imply it is detecting the even correctly (?)
Not sure about this, looks like I am getting two events: * ZT_EVENT_PULSE_START==14 * 65585 Instead of one ZT_EVENT_WINKFLASH==3 (I assume anyway). -- Brian May <brian@microcomaustralia.com.au>
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