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[asterisk-users] Issues with call dropping


 
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jleed at me.com
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PostPosted: Mon Apr 20, 2015 7:08 am    Post subject: [asterisk-users] Issues with call dropping Reply with quote

Hi guys, have really annoying problem with trunks when I calling over voip provider..


after awhile provider sends INFO packages but for some reason Asterisk doesn’t answer on it.
after 8 packagers provider just drops the call, here is the package

<--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 --->
INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9@192.168.53.9:5060 SIP/2.0
Max-Forwards: 69
To: <sip:4959810128@192.168.53.9>;tag=b3769af4-118b-4467-8c95-042247ff1776
From: <sip:84957774888@192.168.53.1>;tag=3638518512-132845
Call-ID: cfe34652-14c2-4072-9dea-f0b0c30cb15e
CSeq: 2 INFO
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 192.168.53.1:5060;branch=z9hG4bK8da58cede20b91eb54dec15ad27f866c
Contact: <sip:84957774888@192.168.53.1:5060>
Content-Length: 0

192.168.53.1 - operator IP
192.168.53.9 - asterisk IP


Any idea how to fix this?


have 2 Ethernet interfaces:
192.168.1.4 - local network
192.168.53.9 - VOIP Provider network

Im using PJSIP, here is config:

[udp]
type=transport
protocol=udp
bind=192.168.1.4
local_net=10.0.0.0/24
local_net=10.0.1.0/24
local_net=192.168.1.0/24

external_media_address=195.239.8.122
external_signaling_address=195.239.8.122

[udp_B]
type=transport
protocol=udp
bind=192.168.53.9

[10000]
type=endpoint
aors=10000
context=dialmap
disallow=all
allow=alaw,ulaw
transport=udp_B

[10000]
type=aor
contact=sip:192.168.53.1:5060
max_contacts=4


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jcolp at digium.com
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PostPosted: Mon Apr 20, 2015 1:38 pm    Post subject: [asterisk-users] Issues with call dropping Reply with quote

Nick Awesome wrote:
Quote:
Hi guys, have really annoying problem with trunks when I calling over voip provider..


after awhile provider sends INFO packages but for some reason Asterisk doesn’t answer on it.
after 8 packagers provider just drops the call, here is the package

<--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 --->
INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9@192.168.53.9:5060 SIP/2.0
Max-Forwards: 69
To:<sip:4959810128@192.168.53.9>;tag=b3769af4-118b-4467-8c95-042247ff1776
From:<sip:84957774888@192.168.53.1>;tag=3638518512-132845
Call-ID: cfe34652-14c2-4072-9dea-f0b0c30cb15e
CSeq: 2 INFO
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 192.168.53.1:5060;branch=z9hG4bK8da58cede20b91eb54dec15ad27f866c
Contact:<sip:84957774888@192.168.53.1:5060>
Content-Length: 0

Looks like they are using INFO as a keepalive mechanism. Since we don't
answer this it'd be a bug. File an issue on the issue tracker[1].

[1] https://issues.asterisk.org/jira

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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