Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI
Setting up Freeswitch to connect to Microsoft Teams' "Direct Routing" (https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan). General call setup-teardown is successful, but I'm encountering a problem with call transfers if the transfer target is another Microsoft Teams user. The most immediate obvious difference between the transfer to an external party and the transfer to another Teams user is in the Refer-To header of the Refer message on the transfer event. In the case of transfer to an external party, the Refer-To is (phone number redacted to 555-555-1000):
REFER-TO: <sip:+15555551000@sip.pstnhub.microsoft.com:5061;user=phone;transport=tls>
In the case of transfer to another Teams user, the transfer Refer-To: header looks like this:
REFER-TO: <sip:sip.pstnhub.microsoft.com:5061;transport=tls;x-m=8:orgid:13bc82d9-003a-43b8-aad2-8f2f561a9b8d>
When this occurs, the last line in the Freeswitch log for the call is:
[DEBUG] sofia.c:8544 Process REFER to [(null)@sip.pstnhub.microsoft.com]
There is nothing else in the log until this times out, no indication it has entered any dialplan, etc. Even on timeout, it is just the teams client saying that the call has failed, and unless one end hangs up, or reconnects to the failed transfer, there is no action in freeswitch. I believe this to be a bug, but wanted to get an opinion before opening a formal bug report. Regards, Ben Kaufman AltiGen Communications, Inc. STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator@altigen.com, and destroy all copies of this message and any attachments.
It’s failing due to no number to redirect to. This orgid thing must be some info about where it is supposed to be redirected to? This isn’t something that is currently supported, but that doesn’t mean there isn’t a way to add it. Do you have any info about how it should be routed? This would be something I’d be willing to entertain pull requests to add support for. Mike
On Sep 11, 2018, at 2:33 PM, Ben Kaufman <ben.kaufman@altigen.com> wrote:
Setting up Freeswitch to connect to Microsoft Teams' "Direct Routing" (https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan). General call setup-teardown is successful, but I'm encountering a problem with call transfers if the transfer target is another Microsoft Teams user.
The most immediate obvious difference between the transfer to an external party and the transfer to another Teams user is in the Refer-To header of the Refer message on the transfer event.
In the case of transfer to an external party, the Refer-To is (phone number redacted to 555-555-1000):
REFER-TO: <sip:+15555551000@sip.pstnhub.microsoft.com:5061;user=phone;transport=tls>
In the case of transfer to another Teams user, the transfer Refer-To: header looks like this:
REFER-TO: <sip:sip.pstnhub.microsoft.com:5061;transport=tls;x-m=8:orgid:13bc82d9-003a-43b8-aad2-8f2f561a9b8d>
When this occurs, the last line in the Freeswitch log for the call is:
[DEBUG] sofia.c:8544 Process REFER to [(null)@sip.pstnhub.microsoft.com]
There is nothing else in the log until this times out, no indication it has entered any dialplan, etc. Even on timeout, it is just the teams client saying that the call has failed, and unless one end hangs up, or reconnects to the failed transfer, there is no action in freeswitch. I believe this to be a bug, but wanted to get an opinion before opening a formal bug report.
It's possibly the orgid. There's also a Referred-By: header that contains a large amount of encoded (or encrypted? It's not base 64) data: REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061;x-m=8:orgid:f4a7fd03-1c14-454b-b910-2e20b20b2ab7;x-t=a3217d98-07a6-45fc-8c9c-ce427f5ed8ac;x-a=qfNyIy1nfSvIYN63RNEJ1hXKpiLor84ceFqs6iAWusN3wcw5S5tDDiugmb%2b9hOLxcObLtp6oAUPkCDqTCPAGfQtCqvbDlTpTSG7YZHZCSuhvMUB%2fwYVDmbkR1COvm72MtszpjCwwBwkNFxRtJB0hsdiprRrQbvZylULQpIcLQKCxPO8U6Rah7XKM1B8hnJGC66ok3gh9yy0J0q3vCKVwC6w7yhz5PxUVswSkhjqvnXq1MRlF4hTfCIP%2f%2bz34EOQwEowWwXJftnYQkSd1K3tjT5i9VdmiZLg%2bgP0znqk%2fQadRE6QdwuABe9oSALATtl808KuviwbZG7fftIeUi4xsrVkO%2bNfwUseeK9YZ8mxUFd2oStRaGVo4ualIZztAti5rEc179dPnw6vpdZbyZMBGmYOjYB%2bN4AbzQsgXGiTMl%2b5Ujqjozeb0L2%2bx7VAoDdkfa%2buGhkFlLi%2buu3r0gEZo7vdxa44FJYyZFrRRGES1PFgkIB0egbG6HBewkI0ucggNKBlzC4r4fjD8IZVE%2bmovyD95sAehns%2bEuhXsBFek8JXI2vnVlYIYyCLNdcFTq9jyUomMze7%2bNK86UX6W6a75x20srK1f0qERu0VN2%2bVVV4P2CQuXWiyIve%2b82CEzN9T4oO8m0ZgIB3s%2fIK%2boZRDTVWZD1FJ6AdBIS8S7qi1kTmVZds45Sh97vzKHnhETYxe3XMRn78SOP9oYJgyfr814CfTC0EwQzZ8jdjVa0HsXIDu6OsT7fOAwtbdTDovntt9J> I have a Sonus box that does handle the REFER in a manner where the INVITE to establish with this header, as well as the aforementioned x-m-8:orgid:.... parameter in the rURI, but I think carrying things over like that could be done in the dialplan (I think). At this point, it doesn't enter the dialplan. Ben Kaufman ben.kaufman@altigen.com Cloud Operations Manager AltiGen Communications, Inc. -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> On Behalf Of Michael Jerris Sent: Wednesday, September 12, 2018 11:36 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI It’s failing due to no number to redirect to. This orgid thing must be some info about where it is supposed to be redirected to? This isn’t something that is currently supported, but that doesn’t mean there isn’t a way to add it. Do you have any info about how it should be routed? This would be something I’d be willing to entertain pull requests to add support for. Mike
On Sep 11, 2018, at 2:33 PM, Ben Kaufman <ben.kaufman@altigen.com> wrote:
Setting up Freeswitch to connect to Microsoft Teams' "Direct Routing" (https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan). General call setup-teardown is successful, but I'm encountering a problem with call transfers if the transfer target is another Microsoft Teams user.
The most immediate obvious difference between the transfer to an external party and the transfer to another Teams user is in the Refer-To header of the Refer message on the transfer event.
In the case of transfer to an external party, the Refer-To is (phone number redacted to 555-555-1000):
REFER-TO: <sip:+15555551000@sip.pstnhub.microsoft.com:5061;user=phone;transport=tls>
In the case of transfer to another Teams user, the transfer Refer-To: header looks like this:
REFER-TO: <sip:sip.pstnhub.microsoft.com:5061;transport=tls;x-m=8:orgid:13bc82d9-003a-43b8-aad2-8f2f561a9b8d>
When this occurs, the last line in the Freeswitch log for the call is:
[DEBUG] sofia.c:8544 Process REFER to [(null)@sip.pstnhub.microsoft.com]
There is nothing else in the log until this times out, no indication it has entered any dialplan, etc. Even on timeout, it is just the teams client saying that the call has failed, and unless one end hangs up, or reconnects to the failed transfer, there is no action in freeswitch. I believe this to be a bug, but wanted to get an opinion before opening a formal bug report.
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator@altigen.com, and destroy all copies of this message and any attachments.
I’m open to patches that add this functionality. Mike
On Sep 12, 2018, at 3:08 PM, Ben Kaufman <ben.kaufman@altigen.com> wrote:
It's possibly the orgid. There's also a Referred-By: header that contains a large amount of encoded (or encrypted? It's not base 64) data:
REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061;x-m=8:orgid:f4a7fd03-1c14-454b-b910-2e20b20b2ab7;x-t=a3217d98-07a6-45fc-8c9c-ce427f5ed8ac;x-a=qfNyIy1nfSvIYN63RNEJ1hXKpiLor84ceFqs6iAWusN3wcw5S5tDDiugmb%2b9hOLxcObLtp6oAUPkCDqTCPAGfQtCqvbDlTpTSG7YZHZCSuhvMUB%2fwYVDmbkR1COvm72MtszpjCwwBwkNFxRtJB0hsdiprRrQbvZylULQpIcLQKCxPO8U6Rah7XKM1B8hnJGC66ok3gh9yy0J0q3vCKVwC6w7yhz5PxUVswSkhjqvnXq1MRlF4hTfCIP%2f%2bz34EOQwEowWwXJftnYQkSd1K3tjT5i9VdmiZLg%2bgP0znqk%2fQadRE6QdwuABe9oSALATtl808KuviwbZG7fftIeUi4xsrVkO%2bNfwUseeK9YZ8mxUFd2oStRaGVo4ualIZztAti5rEc179dPnw6vpdZbyZMBGmYOjYB%2bN4AbzQsgXGiTMl%2b5Ujqjozeb0L2%2bx7VAoDdkfa%2buGhkFlLi%2buu3r0gEZo7vdxa44FJYyZFrRRGES1PFgkIB0egbG6HBewkI0ucggNKBlzC4r4fjD8IZVE%2bmovyD95sAehns%2bEuhXsBFek8JXI2vnVlYIYyCLNdcFTq9jyUomMze7%2bNK86UX6W6a75x20srK1f0qERu0VN2%2bVVV4P2CQuXWiyIve%2b82CEzN9T4oO8m0ZgIB3s%2fIK%2boZRDTVWZD1FJ6AdBIS8S7qi1kTmVZds45Sh97vzKHnhETYxe3XMRn78SOP9oYJgyfr814CfTC0EwQzZ8jdjVa0HsXIDu6OsT7fOAwtbdTDovntt9J>
I have a Sonus box that does handle the REFER in a manner where the INVITE to establish with this header, as well as the aforementioned x-m-8:orgid:.... parameter in the rURI, but I think carrying things over like that could be done in the dialplan (I think). At this point, it doesn't enter the dialplan.
Ben, Thanks for talking to me about this today, can you file a jira with the details of this at freeswitch.org/jira? Thanks, /b On Wed, Sep 12, 2018 at 2:15 PM, Michael Jerris <mike@jerris.com> wrote:
I’m open to patches that add this functionality.
Mike
On Sep 12, 2018, at 3:08 PM, Ben Kaufman <ben.kaufman@altigen.com> wrote:
It's possibly the orgid. There's also a Referred-By: header that contains a large amount of encoded (or encrypted? It's not base 64) data:
REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061;x-m=8:orgid:f4a7fd03- 1c14-454b-b910-2e20b20b2ab7;x-t=a3217d98-07a6-45fc-8c9c-ce427f5ed8ac;x-a= qfNyIy1nfSvIYN63RNEJ1hXKpiLor84ceFqs6iAWusN3wcw5S5tDDiugmb% 2b9hOLxcObLtp6oAUPkCDqTCPAGfQtCqvbDlTpTSG7YZHZCSuhvMUB% 2fwYVDmbkR1COvm72MtszpjCwwBwkNFxRtJB0hsdiprRrQbvZylULQpIcLQK CxPO8U6Rah7XKM1B8hnJGC66ok3gh9yy0J0q3vCKVwC6w7yhz5PxUVswSkhj qvnXq1MRlF4hTfCIP%2f%2bz34EOQwEowWwXJftnYQkSd1K3tjT5i9VdmiZLg%2bgP0znqk% 2fQadRE6QdwuABe9oSALATtl808KuviwbZG7fftIeUi4xsrVkO% 2bNfwUseeK9YZ8mxUFd2oStRaGVo4ualIZztAti5rEc179dPnw6vpdZbyZMB GmYOjYB%2bN4AbzQsgXGiTMl%2b5Ujqjozeb0L2%2bx7VAoDdkfa%2buGhkFlLi% 2buu3r0gEZo7vdxa44FJYyZFrRRGES1PFgkIB0egbG6HBewkI0ucggNKBlzC 4r4fjD8IZVE%2bmovyD95sAehns%2bEuhXsBFek8JXI2vnVlYIYyCLNdcFTq9jyUomMze7% 2bNK86UX6W6a75x20srK1f0qERu0VN2%2bVVV4P2CQuXWiyIve% 2b82CEzN9T4oO8m0ZgIB3s%2fIK%2boZRDTVWZD1FJ6AdBIS8S7qi1kTmV Zds45Sh97vzKHnhETYxe3XMRn78SOP9oYJgyfr814CfTC0EwQzZ8jdjVa0Hs XIDu6OsT7fOAwtbdTDovntt9J>
I have a Sonus box that does handle the REFER in a manner where the INVITE to establish with this header, as well as the aforementioned x-m-8:orgid:.... parameter in the rURI, but I think carrying things over like that could be done in the dialplan (I think). At this point, it doesn't enter the dialplan.
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
I had a lot of trouble with the act of opening a ticket. It’s open now (FS-11401), however any attempt to attach my log files failed. Additionally, it looks like I’m not able to post an update to the ticket either to explain this. I have logs for this, but when i attempt to upload them, Jira gives an error that the upload was not successful and that "An internal error has occured. Please contact your administrator". The log files are less than 200 KB in size,so it's likely not a size issue. I also have a log from a Sonus SBC that does process the transfer which could be used as a comparison of the full dialog. Even when I log into JIRA, it even gives an error: We've detected a potential problem with JIRA's Dashboard configuration that your administrator can correct. Hide Dashboard Diagnostics: Mismatched URL Hostname JIRA is reporting that it is running on the hostname 'freeswitch.org', which does not match the hostname used to run these diagnostics, 'freeswitch.com'. This is known to cause JIRA to construct URLs using the incorrect hostname, which will result in errors in the dashboard, among other issues. The most common cause of this is the use of a reverse-proxy HTTP server (often Apache or IIS) in front of the application server running JIRA. While this configuration is supported, some additional setup might be necessary in order to ensure that JIRA detects the correct hostname. The following articles describe the issue and the steps you should take to ensure that your web server and app server are configured correctly: * Gadgets do not display correctly after upgrade to JIRA 4.0<http://confluence.atlassian.com/display/JIRAKB/Gadgets+do+not+display+correctly+after+upgrade+to+JIRA+4.0> * Integrating JIRA with Apache<http://confluence.atlassian.com/display/JIRA/Integrating+JIRA+with+Apache> * Integrating JIRA with Apache using SSL<http://confluence.atlassian.com/display/JIRA/Integrating+JIRA+with+Apache+using+SSL> If you believe this diagnosis is in error, or you have any other questions, please contact Atlassian Support<http://support.atlassian.com>. Detailed Error Click here to learn more Ben Kaufman ben.kaufman@altigen.com<mailto:ben.kaufman@altigen.com> Cloud Operations Manager AltiGen Communications, Inc. From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> On Behalf Of Brian West Sent: Wednesday, September 12, 2018 5:15 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI Ben, Thanks for talking to me about this today, can you file a jira with the details of this at freeswitch.org/jira<http://freeswitch.org/jira>? Thanks, /b On Wed, Sep 12, 2018 at 2:15 PM, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote: I’m open to patches that add this functionality. Mike
On Sep 12, 2018, at 3:08 PM, Ben Kaufman <ben.kaufman@altigen.com<mailto:ben.kaufman@altigen.com>> wrote:
It's possibly the orgid. There's also a Referred-By: header that contains a large amount of encoded (or encrypted? It's not base 64) data:
REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061;x-m=8:orgid:f4a7fd03-1c14-454b-b910-2e20b20b2ab7;x-t=a3217d98-07a6-45fc-8c9c-ce427f5ed8ac;x-a=qfNyIy1nfSvIYN63RNEJ1hXKpiLor84ceFqs6iAWusN3wcw5S5tDDiugmb%2b9hOLxcObLtp6oAUPkCDqTCPAGfQtCqvbDlTpTSG7YZHZCSuhvMUB%2fwYVDmbkR1COvm72MtszpjCwwBwkNFxRtJB0hsdiprRrQbvZylULQpIcLQKCxPO8U6Rah7XKM1B8hnJGC66ok3gh9yy0J0q3vCKVwC6w7yhz5PxUVswSkhjqvnXq1MRlF4hTfCIP%2f%2bz34EOQwEowWwXJftnYQkSd1K3tjT5i9VdmiZLg%2bgP0znqk%2fQadRE6QdwuABe9oSALATtl808KuviwbZG7fftIeUi4xsrVkO%2bNfwUseeK9YZ8mxUFd2oStRaGVo4ualIZztAti5rEc179dPnw6vpdZbyZMBGmYOjYB%2bN4AbzQsgXGiTMl%2b5Ujqjozeb0L2%2bx7VAoDdkfa%2buGhkFlLi%2buu3r0gEZo7vdxa44FJYyZFrRRGES1PFgkIB0egbG6HBewkI0ucggNKBlzC4r4fjD8IZVE%2bmovyD95sAehns%2bEuhXsBFek8JXI2vnVlYIYyCLNdcFTq9jyUomMze7%2bNK86UX6W6a75x20srK1f0qERu0VN2%2bVVV4P2CQuXWiyIve%2b82CEzN9T4oO8m0ZgIB3s%2fIK%2boZRDTVWZD1FJ6AdBIS8S7qi1kTmVZds45Sh97vzKHnhETYxe3XMRn78SOP9oYJgyfr814CfTC0EwQzZ8jdjVa0HsXIDu6OsT7fOAwtbdTDovntt9J>
I have a Sonus box that does handle the REFER in a manner where the INVITE to establish with this header, as well as the aforementioned x-m-8:orgid:.... parameter in the rURI, but I think carrying things over like that could be done in the dialplan (I think). At this point, it doesn't enter the dialplan.
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com -- [Image removed by sender.] Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com<mailto:sales@freeswitch.com> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com<mailto:brian@freeswitch.com> Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com<https://www.freeswitch.com/> [Image removed by sender. https://www.facebook.com/signalwireinc?src=email]<https://www.facebook.com/freeswitch>[Image removed by sender. https://twitter.com/freeswitch]<https://twitter.com/freeswitch> STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator@altigen.com, and destroy all copies of this message and any attachments.
Hi All, Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP stream on an inbound call from a carrier so that this isn't audible in the call recording while still retaining DTMF Outofband (RFC2833) in the SIP messages? If not, how could this be achieved? Thanks, Shaun
Try setting rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF and enable the jitterbuffer by setting the channel variable jitterbuffer_msec=60. This is the solution to https://freeswitch.org/jira/browse/FS-4905 On Thu, 13 Sep 2018 at 13:26, Shaun Stokes <shaun.stokes@itec-support.co.uk> wrote:
Hi All,
Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP stream on an inbound call from a carrier so that this isn't audible in the call recording while still retaining DTMF Outofband (RFC2833) in the SIP messages? If not, how could this be achieved?
Thanks,
Shaun _________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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Thanks Steve, we'll give this a try. ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Steven Ayre <steveayre@gmail.com> Sent: 13 September 2018 14:07:03 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? Try setting rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF and enable the jitterbuffer by setting the channel variable jitterbuffer_msec=60. This is the solution to https://freeswitch.org/jira/browse/FS-4905 On Thu, 13 Sep 2018 at 13:26, Shaun Stokes <shaun.stokes@itec-support.co.uk<mailto:shaun.stokes@itec-support.co.uk>> wrote: Hi All, Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP stream on an inbound call from a carrier so that this isn't audible in the call recording while still retaining DTMF Outofband (RFC2833) in the SIP messages? If not, how could this be achieved? Thanks, Shaun _________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
After some research we've found that this may achieve the desired results: <action application="set" data="rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF"/> <action application="set" data="jitterbuffer_msec=60"/> <action application="set" data="sip_jitter_buffer_during_bridge=true"/> We will post back with the results after testing. Sources: http://lists.freeswitch.org/pipermail/freeswitch-users/2016-June/121194.html http://lists.freeswitch.org/pipermail/freeswitch-users/2016-July/121356.html ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Shaun Stokes <shaun.stokes@itec-support.co.uk> Sent: 13 September 2018 09:03:20 To: FreeSWITCH Users Help Subject: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? Hi All, Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP stream on an inbound call from a carrier so that this isn't audible in the call recording while still retaining DTMF Outofband (RFC2833) in the SIP messages? If not, how could this be achieved? Thanks, Shaun
participants (5)
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Ben Kaufman
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Brian West
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Michael Jerris
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Shaun Stokes
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Steven Ayre