Hi I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway. I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up. Here is what I have tested 1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up. Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation? I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active. ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way. Thanks in advance O.
check the line supervision signalling of your GSM and the GXW4104. there are several disconnect signalling used: current disconnect, polarity reversal or busy tones. your FXO port must be set accordingly. On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset@the800group.com> wrote:
Hi
I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway.
I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up.
Here is what I have tested
1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up.
Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation?
I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active.
ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way.
Thanks in advance O.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Thanks Nandy I am not using the GSM device in this test and have tried changing the disconnect signalling without any luck. When I hang up the phones, nothing is recorder in the Freeswitch log (log level 7). Is there a way to trap the disconnect signal by writing a "lua" script or would that info already be written to the log if it was being received?. What I don't understand is that when I use the SIP gateway for the outgoing leg of the call, then the disconnect signal from the two phones is received. But when I use the PSTN line as the outgoing leg, the disconnect signal is not received from either phone? Anyone got any suggestions how I would debug this? Thanks again for your help. O On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
check the line supervision signalling of your GSM and the GXW4104. there are several disconnect signalling used: current disconnect, polarity reversal or busy tones. your FXO port must be set accordingly.
On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset@the800group.com <mailto:ocset@the800group.com>> wrote:
Hi
I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway.
I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up.
Here is what I have tested
1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up.
Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation?
I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active.
ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way.
Thanks in advance O.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
GSM device or not, just determine what line disconnect signalling is used in each port. the GWX gateway must be set to detect this. otherwise, there's no way FS can hangup the call. this signal must be present when using the SIP gateway. also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up first. On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset@the800group.com> wrote:
Thanks Nandy
I am not using the GSM device in this test and have tried changing the disconnect signalling without any luck. When I hang up the phones, nothing is recorder in the Freeswitch log (log level 7). Is there a way to trap the disconnect signal by writing a "lua" script or would that info already be written to the log if it was being received?.
What I don't understand is that when I use the SIP gateway for the outgoing leg of the call, then the disconnect signal from the two phones is received. But when I use the PSTN line as the outgoing leg, the disconnect signal is not received from either phone?
Anyone got any suggestions how I would debug this?
Thanks again for your help. O
On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
check the line supervision signalling of your GSM and the GXW4104. there are several disconnect signalling used: current disconnect, polarity reversal or busy tones. your FXO port must be set accordingly.
On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset@the800group.com> wrote:
Hi
I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway.
I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up.
Here is what I have tested
1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up.
Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation?
I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active.
ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way.
Thanks in advance O.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Nandy Thanks for you help. Just to answer you question first. It makes not difference who hangs up first, the calls never terminate. I have enabled "Polarity Reversal" on the GXW4104 and that has solved the problem. What I don't get is why this would work. I thought Polarity Reversal is old school and how would my mobile phone be doing this or is this being done by the mobile phone provider/carrier? I now have "Enable Current Disconnect", "Enable Tone Disconnect" and "Enable Polarity Reversal" set for good measure even though the "Reverse Polarity" was the one that seemed to fix the problem. Regards O. On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
GSM device or not, just determine what line disconnect signalling is used in each port. the GWX gateway must be set to detect this. otherwise, there's no way FS can hangup the call. this signal must be present when using the SIP gateway.
also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up first.
On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset@the800group.com <mailto:ocset@the800group.com>> wrote:
Thanks Nandy
I am not using the GSM device in this test and have tried changing the disconnect signalling without any luck. When I hang up the phones, nothing is recorder in the Freeswitch log (log level 7). Is there a way to trap the disconnect signal by writing a "lua" script or would that info already be written to the log if it was being received?.
What I don't understand is that when I use the SIP gateway for the outgoing leg of the call, then the disconnect signal from the two phones is received. But when I use the PSTN line as the outgoing leg, the disconnect signal is not received from either phone?
Anyone got any suggestions how I would debug this?
Thanks again for your help. O
On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
check the line supervision signalling of your GSM and the GXW4104. there are several disconnect signalling used: current disconnect, polarity reversal or busy tones. your FXO port must be set accordingly.
On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset@the800group.com <mailto:ocset@the800group.com>> wrote:
Hi
I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway.
I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up.
Here is what I have tested
1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up.
Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation?
I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active.
ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way.
Thanks in advance O.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
It is what your telephone company's equipment does, not your mobile phone provider. On Apr 15, 2012 12:39 PM, "ocset" <ocset@the800group.com> wrote:
Hi Nandy
Thanks for you help.
Just to answer you question first. It makes not difference who hangs up first, the calls never terminate.
I have enabled "Polarity Reversal" on the GXW4104 and that has solved the problem. What I don't get is why this would work. I thought Polarity Reversal is old school and how would my mobile phone be doing this or is this being done by the mobile phone provider/carrier?
I now have "Enable Current Disconnect", "Enable Tone Disconnect" and "Enable Polarity Reversal" set for good measure even though the "Reverse Polarity" was the one that seemed to fix the problem.
Regards O.
On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
GSM device or not, just determine what line disconnect signalling is used in each port. the GWX gateway must be set to detect this. otherwise, there's no way FS can hangup the call. this signal must be present when using the SIP gateway.
also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up first.
On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset@the800group.com> wrote:
Thanks Nandy
I am not using the GSM device in this test and have tried changing the disconnect signalling without any luck. When I hang up the phones, nothing is recorder in the Freeswitch log (log level 7). Is there a way to trap the disconnect signal by writing a "lua" script or would that info already be written to the log if it was being received?.
What I don't understand is that when I use the SIP gateway for the outgoing leg of the call, then the disconnect signal from the two phones is received. But when I use the PSTN line as the outgoing leg, the disconnect signal is not received from either phone?
Anyone got any suggestions how I would debug this?
Thanks again for your help. O
On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
check the line supervision signalling of your GSM and the GXW4104. there are several disconnect signalling used: current disconnect, polarity reversal or busy tones. your FXO port must be set accordingly.
On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset@the800group.com> wrote:
Hi
I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway.
I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up.
Here is what I have tested
1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up.
Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation?
I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active.
ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way.
Thanks in advance O.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
nice to know it worked! old school? there are billions of POTS lines still using it although they're feed by digital exchanges:-) On Mon, Apr 16, 2012 at 12:46 AM, Brian Foster <bdfoster@endigotech.com>wrote:
It is what your telephone company's equipment does, not your mobile phone provider. On Apr 15, 2012 12:39 PM, "ocset" <ocset@the800group.com> wrote:
Hi Nandy
Thanks for you help.
Just to answer you question first. It makes not difference who hangs up first, the calls never terminate.
I have enabled "Polarity Reversal" on the GXW4104 and that has solved the problem. What I don't get is why this would work. I thought Polarity Reversal is old school and how would my mobile phone be doing this or is this being done by the mobile phone provider/carrier?
I now have "Enable Current Disconnect", "Enable Tone Disconnect" and "Enable Polarity Reversal" set for good measure even though the "Reverse Polarity" was the one that seemed to fix the problem.
Regards O.
On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
GSM device or not, just determine what line disconnect signalling is used in each port. the GWX gateway must be set to detect this. otherwise, there's no way FS can hangup the call. this signal must be present when using the SIP gateway.
also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up first.
On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset@the800group.com> wrote:
Thanks Nandy
I am not using the GSM device in this test and have tried changing the disconnect signalling without any luck. When I hang up the phones, nothing is recorder in the Freeswitch log (log level 7). Is there a way to trap the disconnect signal by writing a "lua" script or would that info already be written to the log if it was being received?.
What I don't understand is that when I use the SIP gateway for the outgoing leg of the call, then the disconnect signal from the two phones is received. But when I use the PSTN line as the outgoing leg, the disconnect signal is not received from either phone?
Anyone got any suggestions how I would debug this?
Thanks again for your help. O
On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
check the line supervision signalling of your GSM and the GXW4104. there are several disconnect signalling used: current disconnect, polarity reversal or busy tones. your FXO port must be set accordingly.
On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset@the800group.com> wrote:
Hi
I have successfully managed to get FreeSwitch to bridge an incoming call on a PSTN line to an external mobile phone using a second PSTN line with the help of a GXW4104 FSO gateway.
I am now having a serious issue where the call never gets terminated when the parties hang up. The only way to end the call is to either shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a SIP gateway (PennyTel) instead of the second PSTN line, then it works as expected when both parties hang up.
Here is what I have tested
1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external mobile phone -- call end when both parties hang up. 2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104) -> external mobile phone -- call never end, even when both parties hang up.
Is there a parameter I should be setting to help solve this issue or some code that I should implement to test for this situation?
I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), if the person being called does not end the call on their side, then that leg of the bridge remains active.
ps. The second PSTN line is not a real copper line. It is a PennyTel VOIP line that is managed by the modem and connected to the GXW4104 via phone cable. So, as far as the GXW4104 is concerned, it is just another POTS line. I don't have a second PSTN phone line so can't test any other way.
Thanks in advance O.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Brian Foster
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Nandy Dagondon
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ocset