Asterisk incoming call disconnects on pickup

Asterisk Asterisk Support
1

Hi, i am facing an issue. Incoming call disconnects when i try to answer it. I am receiving a bye event.

0x7fc68000a8a0 – Strict RTP learning after remote address set to: 172.31.46.240:26820
– SIP/21000-000000f4 answered SIP/provider-000000f3
Audio is at 18486
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.76.120.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK5ce2.ea181b73247e83f336dd1381977d3484.0;received=192.76.120.10;rport=5060
Via: SIP/2.0/UDP 10.15.50.8:6050;branch=z9hG4bK5ce2.2f08ba959f6c0de12c3f1661abca24c4.0
Via: SIP/2.0/UDP 10.15.59.4:6000;received=10.15.59.4;rport=6000;branch=z9hG4bKQyDyF74KQNrHj
Record-Route: sip:192.76.120.10;r2=on;lr;ftag=4QZ88r05BZgXa
Record-Route: sip:10.255.0.1;r2=on;lr;ftag=4QZ88r05BZgXa
Record-Route: sip:10.15.50.8:6050;lr;tnx=213.669
From: "923154540496"sip:[923154540496@sip.telnyx.com](mailto:923154540496@sip.telnyx.com);tag=4QZ88r05BZgXa
To: sip:[17182809547@kss-us.query.consul](mailto:17182809547@kss-us.query.consul);tag=as024959aa
Call-ID: e315b660-585a-4c3f-a6bb-7f142a1080ce
CSeq: 17064608 INVITE
Server: Asterisk PBX 16.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[17182809547@3.21.50.37](mailto:17182809547@3.21.50.37:5060)
Content-Type: application/sdp
Require: timer
Content-Length: 499

v=0
o=root 1265232442 1265232442 IN IP4 3.21.50.37
s=Asterisk PBX 16.8.0
c=IN IP4 3.21.50.37
t=0 0
m=audio 18486 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:10
a=maxptime:150
a=ice-ufrag:5b1d8ba7553ca9f41e3266b93d221808
a=ice-pwd:29c7a19651791b2006f445c837dcd685
a=candidate:Hac1f2ef0 1 UDP 2130706431 172.31.46.240 18486 typ host
a=candidate:Hac1f2ef0 2 UDP 2130706430 172.31.46.240 18487 typ host
a=sendrecv

<------------>
– Channel SIP/21000-000000f4 joined ‘simple_bridge’ basic-bridge <6daf2259-7d5c-4b65-94b8-dffb6998b3b1>
– Channel SIP/provider-000000f3 joined ‘simple_bridge’ basic-bridge <6daf2259-7d5c-4b65-94b8-dffb6998b3b1>

Bridge 6daf2259-7d5c-4b65-94b8-dffb6998b3b1: switching from simple_bridge technology to native_rtp
Locally RTP bridged ‘SIP/provider-000000f3’ and ‘SIP/21000-000000f4’ in stack

<— SIP read from UDP:192.76.120.10:5060 —>
ACK sip:17182809547@3.21.50.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK5ce2.bfe990fcc58dcb94b565c6b399c348d2.0
Via: SIP/2.0/UDP 10.15.50.8:6050;branch=z9hG4bK5ce2.2bbae36bc6261e3dc9823c1eaafe64ea.0
v:SIP/2.0/UDP 10.15.59.4:6000;received=10.15.59.4;rport=6000;branch=z9hG4bKtSS8mrQyegU9m
Max-Forwards:68
f:"923154540496"sip:[923154540496@sip.telnyx.com](mailto:923154540496@sip.telnyx.com);tag=4QZ88r05BZgXa
t:sip:[17182809547@kss-us.query.consul](mailto:17182809547@kss-us.query.consul);tag=as024959aa
i:e315b660-585a-4c3f-a6bb-7f142a1080ce
CSeq:17064608 ACK
m:sip:[mod_sofia@10.15.59.4](mailto:mod_sofia@10.15.59.4:6000)
l:0

<------------->
— (11 headers 0 lines) —

<— SIP read from UDP:192.76.120.10:5060 —>
BYE sip:17182809547@3.21.50.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK4ce2.6dc0eeb3b35793b9f6e44f663c4db8d3.0
Via: SIP/2.0/UDP 10.15.50.8:6050;branch=z9hG4bK4ce2.d579604f72ea614532e8d65d03b48066.0
v:SIP/2.0/UDP 10.15.59.4:6000;received=10.15.59.4;rport=6000;branch=z9hG4bKU2j1pK81BSHvg
Max-Forwards:68
f:"923154540496"sip:[923154540496@sip.telnyx.com](mailto:923154540496@sip.telnyx.com);tag=4QZ88r05BZgXa
t:sip:[17182809547@kss-us.query.consul](mailto:17182809547@kss-us.query.consul);tag=as024959aa
i:e315b660-585a-4c3f-a6bb-7f142a1080ce
CSeq:17064609 BYE
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,NOTIFY
k:timer,path,replaces
Reason:Q.850;cause=88;text=“INCOMPATIBLE_DESTINATION”
l:0

2

You are receiving a BYE from Telnyx with a reason text of “INCOMPATIBLE_DESTINATION”. Why that is isn’t clear, you’d likely need to reach out to them to see if they can explain why.

3

Thanks. I will try to contact them and see if they can fix it.

4

INCOMPATIBLE_DESTINATION is an error from Freeswitch.
Have you removed SIP messages before paste here? I didn’t saw the codec negotiation between your server and provider.

Try enable all codecs on your peer conf or confirm you have the right one enable on your devide/softphone.

5

On revisiting it may be because you have ICE enabled for them. You should disable this. They may not support it.

1 Like
6

I i disable icesupport in sip.conf then incoming is working fine. But now there is no audio on outgoing call.
Previously, outgoing was working perfectly with audio.

7

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