Incoming call issue error 401 No matching peer for

Hi everyone,

I have the folowing issue with incoming calls.
Sometimes it works and most of the time I got the following error :

[2018-03-16 10:00:13] VERBOSE[8423] chan_sip.c: — (20 headers 16 lines) —
[2018-03-16 10:00:13] VERBOSE[8423] chan_sip.c: Sending to 83.136.163.72:5060 (no NAT)
[2018-03-16 10:00:13] VERBOSE[8423][C-0000001b] chan_sip.c: Sending to 83.136.163.72:5060 (no NAT)
[2018-03-16 10:00:13] VERBOSE[8423][C-0000001b] chan_sip.c: Using INVITE request as basis request - 7104e8ee00c25def[email protected]
[2018-03-16 10:00:13] VERBOSE[8423][C-0000001b] chan_sip.c: No matching peer for ‘337XXXXXX’ from ‘83.136.163.72:5060’
[2018-03-16 10:00:13] VERBOSE[8423][C-0000001b] chan_sip.c:
<— Reliably Transmitting (no NAT) to 83.136.163.72:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 83.136.163.72;branch=z9hG4bK75f9.59273806.0;received=83.136.163.72
Via: SIP/2.0/UDP 192.168.21.62;branch=z9hG4bK75f9.2dc49671.0
Via: SIP/2.0/UDP 192.168.21.62;branch=z9hG4bK75f9.1dc49671.0
Via: SIP/2.0/UDP 192.168.21.155:5060;branch=z9hG4bK6f198a65
From: “3378XXXXX” sip:[email protected];tag=bb2c5af936
To: sip:[email protected];tag=as43ebf03e
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(13.18.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5c8d232b"
Content-Length: 0

<------------>
[2018-03-16 10:00:13] VERBOSE[8423][C-0000001b] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

My trunk :
nat=no
accountcode=334XXXXXX
host=keyyo.net
type=peer
trunkname=keyyo.net
username=334XXXXXX
secret=XXXXXXXXXX
disallow=all
allow=alaw,ulaw
hasiax=no
hassip=yes
registersip=yes
trunkstyle=voip
insecure=port,invite
hasexten=no
qualify=yes
qualifyfreq=20
autoframing=yes
fromuser=334XXXXXX

user details :
secret=XXXXXXXX
type=user
context=from-trunk
qualify=no
host=dynamic

I tried with nat=yes but same issue

Can someone give me a little help ?

Thanks

It sounds like you have an extension defined in your PJ-SIP settings, but you are calling it on the Chan-SIP port (5060). You can either try the other SIP driver port (typically 5160, but your configuration may be different) or you can delete the PJ-SIP extension and recreate it as Chan-SIP.

This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.