I have captured the packets at both the endpoints: Freeswitch does not receive RTP packets, logs: http://pasted.co/a1a50e4e where, when I make call from a registered SIP account configured at a softphone, I clearly hear sound form ivr-demo sound. GSM VOIP Gateway is sending RTP packets: http://pastebin.com/0xmBB8rh Codecs settings at both the ends is PCMA and PCMU only Thanks, Deepika <http://pastebin.com/0xmBB8rh> On Wed, Jun 8, 2016 at 1:00 PM, Benjamin Cropley <benjamin.cropley@gmail.com
wrote:
Deepika,
Have you tried doing packet capture and looking at the RTP data to be sure audio is 100% being transmitted between the two endpoints?
By that I mean..
- Do a packet capture on FreeSWITCH server and check RTP is sent and received - Do the same thing, but from the GSM gateway (if possible) and check the same.
p.s. your log file is truncated on the longer lines :(
Ben
On Wed, Jun 8, 2016 at 5:35 AM, Deepika Yadav <deepikay@iiitd.ac.in> wrote:
Yeah, I removed GSM still no audio but only noise
Codecs PCMU and PCMA seems to be matched
Here are the logs :
On Tue, Jun 7, 2016 at 9:13 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
Delete gsm from.codecs (both inbound and outbound) and try again.
Also, activate debug on console, so you can see more of what happens.
Read the troubleshooting and debugging techniques from www.freeswitch.org/confluence
sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT Il 07/Giu/2016 17:32, "Deepika Yadav" <deepikay@iiitd.ac.in> ha scritto:
Hi,
I want to make calls forwarded from GSM gateway to Freeswitch extension
When I dial a 10- digit mobile number of a sim configured in the GSM gateway in a way that it forwards the incoming call to freeswitch extension. I do not hear anything except for some background noise
Freeswitch codec configuration :
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
GSM- Gateway Codec settings :
priority 1 - G.711 u-law priority 2 - G.711 a-law
Dialplan in public context :
<include> <extension name="public_did"> <condition field="destination_number" expression="^(9811806537)$"> <action application="transfer" data="5000 XML default"/> </condition> </extension> </include>
Freeswitch logs:
2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055 New Channel sofia/external/00919716517818@192.168.20.36:5060 [33e4e958-2c6e-11e6-9af0-278859dd14e6] 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing 00919716517818 <00919716517818>->9811806537 in context public 2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861 Transfer sofia/external/00919716517818@192.168.20.36:5060 to XML[5000@default] 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing 00919716517818 <00919716517818>->5000 in context default 2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/00919716517818@192.168.20.36:5060! 2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268 Channel [sofia/external/00919716517818@192.168.20.36:5060] has been answered 2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto Changing port from 192.168.20.36:31012 to 103.25.231.101:31012 2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup sofia/external/ 00919716517818@192.168.20.36:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1641 Session 262 (sofia/external/00919716517818@192.168.20.36:5060) Ended 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/00919716517818@192.168.20.36:5060 [CS_DESTROY]
Regards, Deepika
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_________________________________________________________________________ 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
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