how to transcode from g.729 to g.711a ?
I'm attempting to call from a SIP phone (g.729) to the SIP gateway that connects g.711a to the PSTN. It's resulting in "SERVICE UNAVAILABLE" .... If I switch my SIP phone to use g.711a, everything works fine. Any sample on how to transcode 9.729 from my SIP phone to the freeswitch server that would then transcode from g.729 to g.711a to connect to the SIP gateway? I've tried g.729 from my SIP phone to call 1000 and 5000 on the freeswitch server and that worked. And I've bought license for g.729. Thanks, Jeff. ------------------------------------------------------------------------ INVITE sip:+6568465283@155.161.200.188 SIP/2.0 Via: SIP/2.0/TCP 155.161.11.194:54927;rport;branch=z9hG4bKPjDNkdtN2t3.xjdgQ-78gpmFRuBhnjA9Qd;alias Max-Forwards: 70 From: "1001" <sip:1001@155.161.200.188>;tag=x0m7o1ogjzCoro6dBITkX2dB5U.4DX-y To: <sip:+6568465283@155.161.200.188> Contact: <sip:1001@155.161.11.194:54927;transport=TCP;ob> Call-ID: kQZCHnkGqshj5lmFItexEB8zNG1MA4W6 CSeq: 21273 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.4 Content-Type: application/sdp Content-Length: 234 v=0 o=- 3643948404 3643948404 IN IP4 155.161.11.194 s=cpc_med c=IN IP4 155.161.11.194 t=0 0 m=audio 4004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ INVITE sip:68465283@155.161.200.188:5070 SIP/2.0 Via: SIP/2.0/UDP 155.161.200.188:5080;rport;branch=z9hG4bKKX98pj6UB2crp Max-Forwards: 69 From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070> Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:gw+outbound1@155.161.200.188:5080;transport=udp;gw=outbound1> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 231 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1001" <sip:1001@155.161.200.188:5070>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1434929593 1434929594 IN IP4 155.161.200.188 s=FreeSWITCH c=IN IP4 155.161.200.188 t=0 0 m=audio 30028 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Content-Length: 0 ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Retry-After: 1 Content-Length: 0 ------------------------------------------------------------------------ My config ... # vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/> # dialplan/default.xml <extension name="oubout_pstn"> <condition field="destination_number" expression="^\+65(.+)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/outbound1/$1"/> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=true"/> </condition> </extension> # sip_profiles/external.xml <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <settings> <param name="disable-transcoding" value="false"/> ... </settings> # sip_profiless/external/example.xml <gateway name="outbound1"> <param name="realm" value="155.161.200.188:5070"/> <param name="register" value="false"/> <variables> <variable name="absolute_codec_string" value="PCMA" direction="outbound"/> </variables> <param name="reg-id" value="1"/> </gateway>
G.729 support is not coming out of the box, because you need to purchase a G.729 license for your FreeSWITCH box. Without the license, the server can only transparently pass the media. Maybe there are other codecs that your phone supports, like iLBC or G722? They don't require the licensing. On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
I'm attempting to call from a SIP phone (g.729) to the SIP gateway that connects g.711a to the PSTN. It's resulting in "SERVICE UNAVAILABLE" ....
If I switch my SIP phone to use g.711a, everything works fine.
Any sample on how to transcode 9.729 from my SIP phone to the freeswitch server that would then transcode from g.729 to g.711a to connect to the SIP gateway?
I've tried g.729 from my SIP phone to call 1000 and 5000 on the freeswitch server and that worked. And I've bought license for g.729.
Thanks, Jeff.
------------------------------------------------------------------------ INVITE sip:+6568465283@155.161.200.188 SIP/2.0 Via: SIP/2.0/TCP 155.161.11.194:54927;rport;branch=z9hG4bKPjDNkdtN2t3.xjdgQ-78gpmFRuBhnjA9Qd;alias Max-Forwards: 70 From: "1001" <sip:1001@155.161.200.188>;tag=x0m7o1ogjzCoro6dBITkX2dB5U.4DX-y To: <sip:+6568465283@155.161.200.188> Contact: <sip:1001@155.161.11.194:54927;transport=TCP;ob> Call-ID: kQZCHnkGqshj5lmFItexEB8zNG1MA4W6 CSeq: 21273 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.4 Content-Type: application/sdp Content-Length: 234
v=0 o=- 3643948404 3643948404 IN IP4 155.161.11.194 s=cpc_med c=IN IP4 155.161.11.194 t=0 0 m=audio 4004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ INVITE sip:68465283@155.161.200.188:5070 SIP/2.0 Via: SIP/2.0/UDP 155.161.200.188:5080;rport;branch=z9hG4bKKX98pj6UB2crp Max-Forwards: 69 From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070> Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:gw+outbound1@155.161.200.188:5080;transport=udp;gw=outbound1> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 231 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1001" <sip:1001@155.161.200.188:5070>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1434929593 1434929594 IN IP4 155.161.200.188 s=FreeSWITCH c=IN IP4 155.161.200.188 t=0 0 m=audio 30028 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Content-Length: 0
------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Retry-After: 1 Content-Length: 0
------------------------------------------------------------------------
My config ...
# vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>
# dialplan/default.xml <extension name="oubout_pstn"> <condition field="destination_number" expression="^\+65(.+)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/outbound1/$1"/> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=true"/> </condition> </extension>
# sip_profiles/external.xml <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <settings> <param name="disable-transcoding" value="false"/> ... </settings>
# sip_profiless/external/example.xml <gateway name="outbound1"> <param name="realm" value="155.161.200.188:5070"/> <param name="register" value="false"/> <variables> <variable name="absolute_codec_string" value="PCMA" direction="outbound"/> </variables> <param name="reg-id" value="1"/> </gateway>
_________________________________________________________________________ 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
On Mon, Jun 22, 2015 at 5:41 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
G.729 support is not coming out of the box, because you need to purchase a G.729 license for your FreeSWITCH box.
Without the license, the server can only transparently pass the media.
On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote: And I've bought license for g.729.
I paid for the license via freeswitch website https://freeswitch.com/cart.php?a=confproduct&i=0, and g729_info shows that I've 2 channels. Any idea on how to do the transcoding? Thanks, Jeff
the call towards sip:68465283@155.161.200.188:5070 contains only G.729 in SDP offer, so the remote side turned it down. You can use the "absolute_codec_string" variable to enforce the G.711 codec on the other side. Also, late negotiation is enabled by default in the SIP profiles. You may want to disable it, so that each leg negotiates the codec more independently. On Mon, Jun 22, 2015 at 12:03 PM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
On Mon, Jun 22, 2015 at 5:41 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
G.729 support is not coming out of the box, because you need to purchase a G.729 license for your FreeSWITCH box.
Without the license, the server can only transparently pass the media.
On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote: And I've bought license for g.729.
I paid for the license via freeswitch website https://freeswitch.com/cart.php?a=confproduct&i=0, and g729_info shows that I've 2 channels.
Any idea on how to do the transcoding?
Thanks, Jeff
_________________________________________________________________________ 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
You cannot bypass media if you want to transcode. Neither you can proxy media. Delete this two lines: => <action application="set" data="proxy_media=false"/> => <action application="set" data="bypass_media=true"/> -giovanni On Mon, Jun 22, 2015 at 12:11 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
the call towards sip:68465283@155.161.200.188:5070 contains only G.729 in SDP offer, so the remote side turned it down. You can use the "absolute_codec_string" variable to enforce the G.711 codec on the other side.
Also, late negotiation is enabled by default in the SIP profiles. You may want to disable it, so that each leg negotiates the codec more independently.
On Mon, Jun 22, 2015 at 12:03 PM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
On Mon, Jun 22, 2015 at 5:41 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
G.729 support is not coming out of the box, because you need to purchase a G.729 license for your FreeSWITCH box.
Without the license, the server can only transparently pass the media.
On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote: And I've bought license for g.729.
I paid for the license via freeswitch website https://freeswitch.com/cart.php?a=confproduct&i=0, and g729_info shows that I've 2 channels.
Any idea on how to do the transcoding?
Thanks, Jeff
_________________________________________________________________________ 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
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
ah, of course. I didn't go that far down the message :-) On Mon, Jun 22, 2015 at 12:16 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
You cannot bypass media if you want to transcode. Neither you can proxy media.
Delete this two lines:
=> <action application="set" data="proxy_media=false"/> => <action application="set" data="bypass_media=true"/>
-giovanni
On Mon, Jun 22, 2015 at 12:11 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
the call towards sip:68465283@155.161.200.188:5070 contains only G.729 in SDP offer, so the remote side turned it down. You can use the "absolute_codec_string" variable to enforce the G.711 codec on the other side.
Also, late negotiation is enabled by default in the SIP profiles. You may want to disable it, so that each leg negotiates the codec more independently.
On Mon, Jun 22, 2015 at 12:03 PM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
On Mon, Jun 22, 2015 at 5:41 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
G.729 support is not coming out of the box, because you need to purchase a G.729 license for your FreeSWITCH box.
Without the license, the server can only transparently pass the media.
On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote: And I've bought license for g.729.
I paid for the license via freeswitch website https://freeswitch.com/cart.php?a=confproduct&i=0, and g729_info shows that I've 2 channels.
Any idea on how to do the transcoding?
Thanks, Jeff
_________________________________________________________________________ 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
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ 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
On Mon, Jun 22, 2015 at 6:25 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
ah, of course. I didn't go that far down the message :-)
On Mon, Jun 22, 2015 at 12:16 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
You cannot bypass media if you want to transcode. Neither you can proxy media.
Delete this two lines:
=> <action application="set" data="proxy_media=false"/> => <action application="set" data="bypass_media=true"/>
-giovanni
On Mon, Jun 22, 2015 at 12:11 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
the call towards sip:68465283@155.161.200.188:5070 contains only G.729 in SDP offer, so the remote side turned it down. You can use the "absolute_codec_string" variable to enforce the G.711 codec on the other side.
Also, late negotiation is enabled by default in the SIP profiles. You may want to disable it, so that each leg negotiates the codec more independently.
On Mon, Jun 22, 2015 at 12:03 PM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
On Mon, Jun 22, 2015 at 5:41 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
G.729 support is not coming out of the box, because you need to purchase a G.729 license for your FreeSWITCH box.
Without the license, the server can only transparently pass the media.
On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote: And I've bought license for g.729.
I paid for the license via freeswitch website https://freeswitch.com/cart.php?a=confproduct&i=0, and g729_info shows that I've 2 channels.
Any idea on how to do the transcoding?
I tried and it still does work. Any simple sample to follow for transcoding g.729 to g.711a ? Here's my modfied sip_profiles/external/example.xml <gateway name="outbound1"> <param name="realm" value="155.161.200.188:5070"/> <param name="register" value="false"/> <param name="inbound-late-negotiation" value="false"/> <variables> <variable name="absolute_codec_string" value="PCMA" direction="outbound"/> </variables> <param name="reg-id" value="1"/> </gateway> I'm seeing this error ... 2015-06-25 13:48:10.580918 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1001@155.161.200.188 [4777a3ee-e25a-481b-91ad-52eaa247a624] 2015-06-25 13:48:10.640922 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->+6568465283 in context default 2015-06-25 13:48:10.640922 [NOTICE] switch_channel.c:1075 New Channel sofia/external/68465283 [9c4747f3-6a36-4933-b158-bcead6bd0ed9] 2015-06-25 13:48:10.660920 [NOTICE] sofia.c:7543 Hangup sofia/external/68465283 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/external/68465283) Ended 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/68465283 [CS_DESTROY] 2015-06-25 13:48:10.660920 [INFO] mod_dptools.c:3246 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2015-06-25 13:48:10.660920 [NOTICE] switch_channel.c:4747 Hangup sofia/internal/1001@155.161.200.188 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/1001@155.161.200.188) Ended 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001@155.161.200.188 [CS_DESTROY] Thanks, Jeff
better set the variable absolute_codec_string in your dialplan before bridging: https://wiki.freeswitch.org/wiki/Variable_absolute_codec_string On Thu, Jun 25, 2015 at 7:48 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
On Mon, Jun 22, 2015 at 6:25 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
ah, of course. I didn't go that far down the message :-)
On Mon, Jun 22, 2015 at 12:16 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
You cannot bypass media if you want to transcode. Neither you can proxy media.
Delete this two lines:
=> <action application="set" data="proxy_media=false"/> => <action application="set" data="bypass_media=true"/>
-giovanni
On Mon, Jun 22, 2015 at 12:11 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
the call towards sip:68465283@155.161.200.188:5070 contains only G.729 in SDP offer, so the remote side turned it down. You can use the "absolute_codec_string" variable to enforce the G.711 codec on the other side.
Also, late negotiation is enabled by default in the SIP profiles. You may want to disable it, so that each leg negotiates the codec more independently.
On Mon, Jun 22, 2015 at 12:03 PM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
On Mon, Jun 22, 2015 at 5:41 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
G.729 support is not coming out of the box, because you need to purchase a G.729 license for your FreeSWITCH box.
Without the license, the server can only transparently pass the media.
On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote: And I've bought license for g.729.
I paid for the license via freeswitch website https://freeswitch.com/cart.php?a=confproduct&i=0, and g729_info shows that I've 2 channels.
Any idea on how to do the transcoding?
I tried and it still does work. Any simple sample to follow for transcoding g.729 to g.711a ?
Here's my modfied sip_profiles/external/example.xml <gateway name="outbound1"> <param name="realm" value="155.161.200.188:5070"/> <param name="register" value="false"/> <param name="inbound-late-negotiation" value="false"/> <variables> <variable name="absolute_codec_string" value="PCMA" direction="outbound"/> </variables> <param name="reg-id" value="1"/> </gateway>
I'm seeing this error ... 2015-06-25 13:48:10.580918 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1001@155.161.200.188 [4777a3ee-e25a-481b-91ad-52eaa247a624] 2015-06-25 13:48:10.640922 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->+6568465283 in context default 2015-06-25 13:48:10.640922 [NOTICE] switch_channel.c:1075 New Channel sofia/external/68465283 [9c4747f3-6a36-4933-b158-bcead6bd0ed9] 2015-06-25 13:48:10.660920 [NOTICE] sofia.c:7543 Hangup sofia/external/68465283 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/external/68465283) Ended 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/68465283 [CS_DESTROY] 2015-06-25 13:48:10.660920 [INFO] mod_dptools.c:3246 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2015-06-25 13:48:10.660920 [NOTICE] switch_channel.c:4747 Hangup sofia/internal/1001@155.161.200.188 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/1001@155.161.200.188) Ended 2015-06-25 13:48:10.660920 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001@155.161.200.188 [CS_DESTROY]
Thanks, Jeff
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On Thu, Jun 25, 2015 at 1:59 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
better set the variable absolute_codec_string in your dialplan before bridging: https://wiki.freeswitch.org/wiki/Variable_absolute_codec_string
On Thu, Jun 25, 2015 at 7:48 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
I tried and it still does work. Any simple sample to follow for transcoding g.729 to g.711a ?
Excellent! That works!!! Thank you! Jeff
participants (3)
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Giovanni Maruzzelli
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Jeff Chua
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Stanislav Sinyagin