Hi,

If you check following log , There is a almost delay of 7 sec. Please let me know what may be wrong. I have manually hide my client IP and phone number in logs.

2016-05-08 06:18:30.288192 [DEBUG] switch_ivr_originate.c:3607 Originate Resulted in Success: [sofia/sipinterface_1/xxxxxxxxxxx@sbc.voxbeam.com]

2016-05-08 06:18:30.288192 [DEBUG] sofia.c:6858 Channel sofia/sipinterface_1/nuz54@XXX.82.43.202:5060 entering state [early][183]

2016-05-08 06:18:30.288192 [DEBUG] switch_core_media.c:9483 sofia/sipinterface_1/xxxxxxxxxxx@sbc.voxbeam.com PAUSE Jitterbuffer

2016-05-08 06:18:30.288192 [DEBUG] switch_core_media.c:9483 sofia/sipinterface_1/nul554@xxx.xx.43.202:5060 PAUSE Jitterbuffer

2016-05-08 06:18:30.288192 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 4849

2016-05-08 06:18:30.288192 [DEBUG] switch_ivr_bridge.c:1594 (sofia/sipinterface_1/xxxxxxxxxxx@sbc.voxbeam.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA

2016-05-08 06:18:30.288192 [DEBUG] switch_core_state_machine.c:543 (sofia/sipinterface_1/xxxxxxxxxxxxx@sbc.voxbeam.com) Running State Change CS_EXCHANGE_MEDIA

2016-05-08 06:18:30.288192 [DEBUG] switch_core_state_machine.c:612 (sofia/sipinterface_1/xxxxxxxxxx@sbc.voxbeam.com) State EXCHANGE_MEDIA

2016-05-08 06:18:30.288192 [DEBUG] mod_sofia.c:628 SOFIA EXCHANGE_MEDIA

2016-05-08 06:18:30.308197 [DEBUG] switch_rtp.c:6694 Correct audio ip/port confirmed.

2016-05-08 06:18:30.728192 [DEBUG] switch_rtp.c:6694 Correct audio ip/port confirmed.

2016-05-08 06:18:37.168193 [NOTICE] switch_channel.c:1104 New Channel sofia/sipinterface_1/51841@173.82.43.202:5060 [GWbegcciibgkinub3864822@173.82.43.202]

2016-05-08 06:18:37.168193 [DEBUG] switch_core_state_machine.c:543 (sofia/sipinterface_1/51841@xxx.82.43.202:5060) Running State Change CS_NEW

2016-05-08 06:18:37.168193 [DEBUG] sofia.c:9374 sofia/sipinterface_1/51841@xxx.82.43.202:5060 receiving invite from 173.82.43.202:5060 version: 1.7.0 64bit

2016-05-08 06:18:37.168193 [DEBUG] sofia.c:9486 IP xxx.82.43.202 Approved by acl "trusted[]". Access Granted.

2016-05-08 06:18:37.168193 [DEBUG] sofia.c:6858 Channel sofia/sipinterface_1/51841@xxx.82.43.202:5060 entering state [received][100]

2016-05-08 06:18:37.168193 [DEBUG] sofia.c:6868 Remote SDP:

Thanks,

Abhishake

On Sun, May 8, 2016 at 1:09 AM, Abhishake Agarwal <abhishake.agarwal@gmail.com> wrote:

Hi,

I am using freeswitch 1.6 with Kazoo just for routing calls to external provider

I added following config in freeswitch.xml

<X-PRE-PROCESS cmd="set" data="ignore_early_media=ring_ready"/>
<X-PRE-PROCESS cmd="set" data="instant_ringback=true"/>
<X-PRE-PROCESS cmd="set" data="return_ring_ready=true"/>
<X-PRE-PROCESS cmd="set" data="ringback=%(2000,4000,440.0,480.0)"/>

Still there is a delay of 8 sec from dialing to ringback. But when I see logs I see logs instantly when I hit dial button.

I want something like whatsapp calls, as we press dial button, there should be instant ring.

Thanks,
Abhishake

On Wed, May 4, 2016 at 8:42 PM, Michael Jerris <mike@jerris.com> wrote:
What you describe will lead to a ring tone followed by a busy signal in some situations. We have had people try this before and every time they do not like the results.

Mike
On May 4, 2016, at 5:13 AM, Abhishake Agarwal <abhishake.agarwal@gmail.com> wrote:

Hi,

When user dial a number, early media is active only after 7-10 sec. even after early media is active, actual ring comes after 5-10 sec, till that time caller does not hear anything. I am need of feature, so that user can hear at least progress beep sound, till caller does not hear a actual ring from b leg.

Thanks,
Abhishake

On Wed, May 4, 2016 at 10:06 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
What must do FreeSwitch after 180 ringing or 183 early media messages?

On Wed, May 4, 2016, 06:41 Sergey Safarov <s.safarov@gmail.com> wrote:
Look at FS-9118

ср, 4 мая 2016 г. в 4:31, Abhishake Agarwal <abhishake.agarwal@gmail.com>:
Hi,

I need beep ring until 180 ring is not received or before early media. Please help me resolving this issue.

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