Hi all, I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch. I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN. When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing... I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log. I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up) I've googled the list, and I've found several people with a similar problem but no solution... That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822 Thank you very much in advance.
This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives. Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call. If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call. Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers. On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella@gmail.com> wrote:
Hi all,
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.
When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...
I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.
I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)
I've googled the list, and I've found several people with a similar problem but no solution...
That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822
Thank you very much in advance.
_______________________________________________ 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/ 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
I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly. Regards, Ognjen (sekil) On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives.
Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call.
If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call.
Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers.
On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella@gmail.com> wrote:
Hi all,
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.
When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...
I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.
I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)
I've googled the list, and I've found several people with a similar problem but no solution...
That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822
Thank you very much in advance.
_______________________________________________ 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/
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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Ok, as discussed with Tony on IRC channel I followed his directions which lead to a successfull outcome (like it always does I might add :). One has to use tone_detect app in FreeSWITCH dialplan in order to check for busy tones coming from the PSTN side and if matched fire a hangup application. This is the snippet of my test dp that does the trick (from extension Local_extensions in default.xml): <anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <anti-action application="bridge" data=" user/${dialed_extension}@${domain_name}"/<user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/>
This means that FS will listen to freq of 425 Hz and wait for 4 positive detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 Hz is the freq telco here uses; for other countries I suggest getting the ITU world tones pdf file and check there): 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 1/4 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 2/4 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 3/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 4/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() TONE busy DETECTED 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] Regards, Ognjen On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <oseslija@gmail.com> wrote:
I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly.
Regards, Ognjen (sekil) On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives.
Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call.
If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call.
Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers.
On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella@gmail.com>wrote:
Hi all,
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.
When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...
I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.
I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)
I've googled the list, and I've found several people with a similar problem but no solution...
That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822
Thank you very much in advance.
_______________________________________________ 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/
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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Scott, I imagined that it could be an OpenZap problem, but I didn't find an OpenZap mailing list, so I sent the email to FS list. Do you know where can I find more information about OpenZap hardware support and developement status (I have special interest in Loop Start)?? Anthony and Ognjen, I've tried tone detection and thanks to that FS is detecting hung up, but I faced the problem that tone detector answer the call... That's my dialplan: <extension name="extension_name"> <condition field="destination_number" expression="^919999999$"> <action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <action application="bridge" data="sofia/internal/1003%${server-domain-name}, sofia/internal/1004%${server-domain-name}"/> </condition> </extension> When I receive a call from PSTN, tone detection answer the call (the caller hears only one first tone and then hears "nothing" until I pick up the call on softphone). So, I think that tone detection solution does not resolve my problem... Is there any other possibility to detect hang up without answering the call (using Loop Start signaling) or have we to wait until OpenZap is completely developed? Thanks in advance. On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija <oseslija@gmail.com> wrote:
Ok, as discussed with Tony on IRC channel I followed his directions which lead to a successfull outcome (like it always does I might add :).
One has to use tone_detect app in FreeSWITCH dialplan in order to check for busy tones coming from the PSTN side and if matched fire a hangup application. This is the snippet of my test dp that does the trick (from extension Local_extensions in default.xml):
<anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <anti-action application="bridge" data=" user/${dialed_extension}@${domain_name}"/<user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/>
This means that FS will listen to freq of 425 Hz and wait for 4 positive detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 Hz is the freq telco here uses; for other countries I suggest getting the ITU world tones pdf file and check there):
2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 1/4 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 2/4 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 3/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 4/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() TONE busy DETECTED 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]
Regards, Ognjen
On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <oseslija@gmail.com>wrote:
I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly.
Regards, Ognjen (sekil) On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives.
Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call.
If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call.
Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers.
On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella@gmail.com>wrote:
Hi all,
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.
When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...
I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.
I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)
I've googled the list, and I've found several people with a similar problem but no solution...
That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822
Thank you very much in advance.
_______________________________________________ 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/
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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When call comes in from Openzap, tone_detect app does pre_answer of a call cause it's need media to start detecting tones in the first place. This behaviour is something that I see on calls inside my telco when calling from analogue lines. I don't think this is a big of deal because ringback provided by FS will make caller understand that the call is still in progress. One can make its own ringback to sound exactly the same as telco's. I don't think that we'll ever make POTS behave like digital protocols do. Regards, Ognjen On Wed, Jan 21, 2009 at 10:36 AM, Scott Ellis <scott.ellis@novatex.com.au>wrote:
I had a similar problem, you can use <action application="set" data="ringback=${au-ring}"/> (I added an "au" ring definition to my vars.xml file)
To get what you want.
I also had a problem that you get two rings, then an answer then to the system generated ring tone, which was confusing some of our (not to bright) callers.
As we don't use callerID I turned that flag off in the openzap.conf.xml file - I thought that this would do what I wanted (answer the instant the call is detected), but the change in the config file does not make it all the way down to the point where it takes action. At this point I hacked the code to get what I wanted. I have to create a JIRA entry with the details yet.
As far as I understand, this is the right place for OpenZap, as it is a product of the FS project.
Scott
Tomás wrote:
Scott, I imagined that it could be an OpenZap problem, but I didn't find an OpenZap mailing list, so I sent the email to FS list. Do you know where can I find more information about OpenZap hardware support and developement status (I have special interest in Loop Start)??
Anthony and Ognjen, I've tried tone detection and thanks to that FS is detecting hung up, but I faced the problem that tone detector answer the call...
That's my dialplan:
<extension name="extension_name"> <condition field="destination_number" expression="^919999999$"> <action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <action application="bridge" data="sofia/internal/1003%${server-domain-name}, sofia/internal/1004%${server-domain-name}"/> </condition> </extension>
When I receive a call from PSTN, tone detection answer the call (the caller hears only one first tone and then hears "nothing" until I pick up the call on softphone).
So, I think that tone detection solution does not resolve my problem... Is there any other possibility to detect hang up without answering the call (using Loop Start signaling) or have we to wait until OpenZap is completely developed?
Thanks in advance.
On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija <oseslija@gmail.com>wrote:
Ok, as discussed with Tony on IRC channel I followed his directions which lead to a successfull outcome (like it always does I might add :).
One has to use tone_detect app in FreeSWITCH dialplan in order to check for busy tones coming from the PSTN side and if matched fire a hangup application. This is the snippet of my test dp that does the trick (from extension Local_extensions in default.xml):
<anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <anti-action application="bridge" data=" user/${dialed_extension}@${domain_name}"/<user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/>
This means that FS will listen to freq of 425 Hz and wait for 4 positive detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 Hz is the freq telco here uses; for other countries I suggest getting the ITU world tones pdf file and check there):
2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 1/4 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 2/4 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 3/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 4/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() TONE busy DETECTED 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]
Regards, Ognjen
On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <oseslija@gmail.com>wrote:
I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly.
Regards, Ognjen (sekil) On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives.
Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call.
If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call.
Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers.
On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella@gmail.com>wrote:
Hi all,
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.
When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...
I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.
I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)
I've googled the list, and I've found several people with a similar problem but no solution...
That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822
Thank you very much in advance.
_______________________________________________ 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/
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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------------------------------
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On Wed, Jan 21, 2009 at 1:49 AM, Ognjen Seslija <oseslija@gmail.com> wrote:
When call comes in from Openzap, tone_detect app does pre_answer of a call cause it's need media to start detecting tones in the first place. This behaviour is something that I see on calls inside my telco when calling from analogue lines. I don't think this is a big of deal because ringback provided by FS will make caller understand that the call is still in progress. One can make its own ringback to sound exactly the same as telco's.
I don't think that we'll ever make POTS behave like digital protocols do.
So true! -MC
Regards, Ognjen
On Wed, Jan 21, 2009 at 10:36 AM, Scott Ellis <scott.ellis@novatex.com.au> wrote:
I had a similar problem, you can use <action application="set" data="ringback=${au-ring}"/> (I added an "au" ring definition to my vars.xml file)
To get what you want.
I also had a problem that you get two rings, then an answer then to the system generated ring tone, which was confusing some of our (not to bright) callers.
As we don't use callerID I turned that flag off in the openzap.conf.xml file - I thought that this would do what I wanted (answer the instant the call is detected), but the change in the config file does not make it all the way down to the point where it takes action. At this point I hacked the code to get what I wanted. I have to create a JIRA entry with the details yet.
As far as I understand, this is the right place for OpenZap, as it is a product of the FS project.
Scott
Tomás wrote:
Scott, I imagined that it could be an OpenZap problem, but I didn't find an OpenZap mailing list, so I sent the email to FS list. Do you know where can I find more information about OpenZap hardware support and developement status (I have special interest in Loop Start)??
Anthony and Ognjen, I've tried tone detection and thanks to that FS is detecting hung up, but I faced the problem that tone detector answer the call...
That's my dialplan:
<extension name="extension_name"> <condition field="destination_number" expression="^919999999$"> <action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <action application="bridge" data="sofia/internal/1003%${server-domain-name}, sofia/internal/1004%${server-domain-name}"/> </condition> </extension>
When I receive a call from PSTN, tone detection answer the call (the caller hears only one first tone and then hears "nothing" until I pick up the call on softphone).
So, I think that tone detection solution does not resolve my problem... Is there any other possibility to detect hang up without answering the call (using Loop Start signaling) or have we to wait until OpenZap is completely developed?
Thanks in advance.
On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija <oseslija@gmail.com> wrote:
Ok, as discussed with Tony on IRC channel I followed his directions which lead to a successfull outcome (like it always does I might add :).
One has to use tone_detect app in FreeSWITCH dialplan in order to check for busy tones coming from the PSTN side and if matched fire a hangup application. This is the snippet of my test dp that does the trick (from extension Local_extensions in default.xml):
<anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/> This means that FS will listen to freq of 425 Hz and wait for 4 positive detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 Hz is the freq telco here uses; for other countries I suggest getting the ITU world tones pdf file and check there):
2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 1/4 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 2/4 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 3/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 4/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() TONE busy DETECTED 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]
Regards, Ognjen
On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <oseslija@gmail.com> wrote:
I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly.
Regards, Ognjen (sekil) On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives.
Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call.
If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call.
Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers.
On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella@gmail.com> wrote:
Hi all,
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.
When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...
I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.
I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)
I've googled the list, and I've found several people with a similar problem but no solution...
That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822
Thank you very much in advance.
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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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On Wed, Jan 21, 2009 at 1:36 AM, Scott Ellis <scott.ellis@novatex.com.au> wrote:
I had a similar problem, you can use <action application="set" data="ringback=${au-ring}"/> (I added an "au" ring definition to my vars.xml file)
To get what you want.
I also had a problem that you get two rings, then an answer then to the system generated ring tone, which was confusing some of our (not to bright) callers.
As we don't use callerID I turned that flag off in the openzap.conf.xml file - I thought that this would do what I wanted (answer the instant the call is detected), but the change in the config file does not make it all the way down to the point where it takes action. At this point I hacked the code to get what I wanted. I have to create a JIRA entry with the details yet.
As far as I understand, this is the right place for OpenZap, as it is a product of the FS project.
At this point there is not a separate mailing list for OpenZAP stuff so here is as good a place as any to ask OZ questions. :) -MC
Scott
Tomás wrote:
Scott, I imagined that it could be an OpenZap problem, but I didn't find an OpenZap mailing list, so I sent the email to FS list. Do you know where can I find more information about OpenZap hardware support and developement status (I have special interest in Loop Start)??
Anthony and Ognjen, I've tried tone detection and thanks to that FS is detecting hung up, but I faced the problem that tone detector answer the call...
That's my dialplan:
<extension name="extension_name"> <condition field="destination_number" expression="^919999999$"> <action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <action application="bridge" data="sofia/internal/1003%${server-domain-name}, sofia/internal/1004%${server-domain-name}"/> </condition> </extension>
When I receive a call from PSTN, tone detection answer the call (the caller hears only one first tone and then hears "nothing" until I pick up the call on softphone).
So, I think that tone detection solution does not resolve my problem... Is there any other possibility to detect hang up without answering the call (using Loop Start signaling) or have we to wait until OpenZap is completely developed?
Thanks in advance.
On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija <oseslija@gmail.com> wrote:
Ok, as discussed with Tony on IRC channel I followed his directions which lead to a successfull outcome (like it always does I might add :).
One has to use tone_detect app in FreeSWITCH dialplan in order to check for busy tones coming from the PSTN side and if matched fire a hangup application. This is the snippet of my test dp that does the trick (from extension Local_extensions in default.xml):
<anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/> <anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/> This means that FS will listen to freq of 425 Hz and wait for 4 positive detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 Hz is the freq telco here uses; for other countries I suggest getting the ITU world tones pdf file and check there):
2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 1/4 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 2/4 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 3/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 4/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() TONE busy DETECTED 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]
Regards, Ognjen
On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <oseslija@gmail.com> wrote:
I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly.
Regards, Ognjen (sekil) On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives.
Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call.
If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call.
Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers.
On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella@gmail.com> wrote:
Hi all,
I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.
When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...
I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.
I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)
I've googled the list, and I've found several people with a similar problem but no solution...
That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822
Thank you very much in advance.
_______________________________________________ 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/
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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participants (5)
-
Anthony Minessale
-
Michael Collins
-
Ognjen Seslija
-
Scott Ellis
-
Tomás