VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
jleed at me.com
Guest
|
 Posted: Tue Jun 30, 2015 8:58 am Post subject: [asterisk-users] Issues with call dropping |
 |
|
May someone help with the sourcecode, trying find where can I manually send response on Received INFO request in PJSIP
ASTERISK-24986 issues opened already more the 2 month and calls from customers still drops. very annoying  maybe some one could help me figure out where Received INFO request dies in source so I could patch it to response 200 OK ?
| Quote: | On 20 Apr 2015, at 15:08, Nick Awesome <jleed@me.com> wrote:
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
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
 |
jcolp at digium.com
Guest
|
| Posted: Tue Jun 30, 2015 9:08 am Post subject: [asterisk-users] Issues with call dropping |
|
|
Nick Awesome wrote:
| Quote: | May someone help with the sourcecode, trying find where can I
manually send response on Received INFO request in PJSIP
ASTERISK-24986 issues opened already more the 2 month and calls from
customers still drops. very annoying maybe some one could help me
figure out where Received INFO request dies in source so I could
patch it to response 200 OK ?
|
INFO support is currently implemented only for DTMF in the
res_pjsip_dtmf_info module. This module is located at
res/res_pjsip_dtmf_info.c
It could be used as a base to implement this, or extended to support it.
Cheers,
--
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
