bypass_media after receiving invite with replaces header
Hello, I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch. However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true. It seems that these variables are not checked when the invite with replaces header is handled. Anyone an idea how to fix this? Best regards, Sjoerd. ----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E‑mail: s.boomstra@telecats.nl |
1.4 is long since EOL. Please try latest release version and confirm this is still an issue.
On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra <S.Boomstra@telecats.nl> wrote:
Hello,
I've got a problem with FreeSwitch version 1.4.26. Our setup is like this:
We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch.
However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true.
It seems that these variables are not checked when the invite with replaces header is handled.
Anyone an idea how to fix this?
Best regards, Sjoerd.
----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E‑mail: s.boomstra@telecats.nl <mailto:s.boomstra@telecats.nl> |
Alternatively, configure the SIP profile to use proxy media by default: <http://wiki.freeswitch.org/wiki/Proxy_Media> <param name="inbound-proxy-media" value="true"/> пт, 18 нояб. 2016 г. в 17:52, Michael Jerris <mike@jerris.com>:
1.4 is long since EOL. Please try latest release version and confirm this is still an issue.
On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra <S.Boomstra@telecats.nl> wrote:
Hello,
I've got a problem with FreeSwitch version 1.4.26. Our setup is like this:
We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch.
However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true.
It seems that these variables are not checked when the invite with replaces header is handled.
Anyone an idea how to fix this?
Best regards, Sjoerd.
----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E‑mail: s.boomstra@telecats.nl |
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Never use proxy media unless there is a specific very good reason to use it. It was originally built for codecs we didn't have proper support for and now has little to no reason to ever use it now that we have video codec and fax support. On Fri, Nov 18, 2016 at 9:00 PM Sergey Safarov <s.safarov@gmail.com> wrote:
Alternatively, configure the SIP profile to use proxy media by default: <http://wiki.freeswitch.org/wiki/Proxy_Media>
<param name="inbound-proxy-media" value="true"/>
пт, 18 нояб. 2016 г. в 17:52, Michael Jerris <mike@jerris.com>:
1.4 is long since EOL. Please try latest release version and confirm this is still an issue.
On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra <S.Boomstra@telecats.nl> wrote:
Hello,
I've got a problem with FreeSwitch version 1.4.26. Our setup is like this:
We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch.
However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true.
It seems that these variables are not checked when the invite with replaces header is handled.
Anyone an idea how to fix this?
Best regards, Sjoerd.
----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E‑mail: s.boomstra@telecats.nl |
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
We will try this issue with the latest version. We have a lot of things attached to FS, so this will take a while (a few weeks). Best regards, Sjoerd. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: vrijdag 18 november 2016 15:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] bypass_media after receiving invite with replaces header 1.4 is long since EOL. Please try latest release version and confirm this is still an issue. On Nov 18, 2016, at 7:09 AM, Sjoerd Boomstra <S.Boomstra@telecats.nl<mailto:S.Boomstra@telecats.nl>> wrote: Hello, I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch. However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true. It seems that these variables are not checked when the invite with replaces header is handled. Anyone an idea how to fix this? Best regards, Sjoerd. ----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E‑mail: s.boomstra@telecats.nl<mailto:s.boomstra@telecats.nl> |
participants (3)
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Michael Jerris
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Sergey Safarov
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Sjoerd Boomstra