Good afternoon,
During two years I have been installing asterisk. Today setting up a new trunk I found an issue that it doesn’t allow me to pass incoming call trough that new trunk. I can make calls but not receive.
The error that I have when I receive a call is:
Before receiving the call I receive an invite from the provider:
<--- SIP read from UDP:199.36.250.60:5060 --->
INVITE sip:0138387b18ea542af7000100620001@98.174.154.133:5060 SIP/2.0
Record-Route: <sip:199.36.250.60:5061;ftag=as6b8a3a35;lr=on>
m: <sip:9492325109@199.36.250.60:5061>
Via: SIP/2.0/UDP 199.36.250.60:5061;branch=z9hG4bK41b4.8e586c86.0
v: SIP/2.0/UDP 10.101.7.24:5060;received=10.101.7.24;branch=z9hG4bK63dfdbfa;rport=5060
Max-Forwards: 69
f: "MANTECON,HECTOR" <sip:9492325109@jiveip.net>;tag=as6b8a3a35
t: <sip:0138387b18ea542af7000100620001@10.101.3.1:5061>
i: 4bfe07e05ec19c1a04e7c3675cec30a1@jiveip.net
CSeq: 102 INVITE
User-Agent: Jive Application Server
Date: Thu, 06 Feb 2014 22:14:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces
X-ref: 570615494_21262505@4.55.3.161
X-jid: pbx24.c1.jiveip.net-1391724857.2444415
Privacy: none
c: application/sdp
l: 320
P-hint: NAT
v=0
o=root 1782689202 1782689202 IN IP4 199.36.250.31
s=Jive
c=IN IP4 199.36.250.31
t=0 0
m=audio 24582 RTP/AVP 0 9 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (20 headers 15 lines) ---
Sending to 199.36.250.60:5060 (NAT)
Sending to 199.36.250.60:5060 (NAT)
Using INVITE request as basis request - 4bfe07e05ec19c1a04e7c3675cec30a1@jiveip.net
Found peer '0138387b18ea542af7000100620001' for '9492325109' from 199.36.250.60:5060
<--- Reliably Transmitting (NAT) to 199.36.250.60:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 199.36.250.60:5061;branch=z9hG4bK41b4.8e586c86.0;received=199.36.250.60;rport=5060
Via: SIP/2.0/UDP 10.101.7.24:5060;received=10.101.7.24;branch=z9hG4bK63dfdbfa;rport=5060
From: "MANTECON,HECTOR" <sip:9492325109@jiveip.net>;tag=as6b8a3a35
To: <sip:0138387b18ea542af7000100620001@10.101.3.1:5061>;tag=as7067db29
Call-ID: 4bfe07e05ec19c1a04e7c3675cec30a1@jiveip.net
CSeq: 102 INVITE
Server: Cisco/SPA303-7.5.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67552883"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '4bfe07e05ec19c1a04e7c3675cec30a1@jiveip.net' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:199.36.250.60:5060 --->
ACK sip:0138387b18ea542af7000100620001@98.174.154.133:5060 SIP/2.0
Via: SIP/2.0/UDP 199.36.250.60:5061;branch=z9hG4bK41b4.8e586c86.0
From: "MANTECON,HECTOR" <sip:9492325109@jiveip.net>;tag=as6b8a3a35
i: 4bfe07e05ec19c1a04e7c3675cec30a1@jiveip.net
To: <sip:0138387b18ea542af7000100620001@10.101.3.1:5061>;tag=as7067db29
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:199.36.250.60:5060 --->
INVITE sip:0138387b18ea542af7000100620001@98.174.154.133:5060 SIP/2.0
Record-Route: <sip:199.36.250.60:5061;ftag=as6b8a3a35;lr=on>
m: <sip:9492325109@199.36.250.60:5061>
Via: SIP/2.0/UDP 199.36.250.60:5061;branch=z9hG4bK51b4.691c11e7.0
v: SIP/2.0/UDP 10.101.7.24:5060;received=10.101.7.24;branch=z9hG4bK6ece369a;rport=5060
Max-Forwards: 69
f: "MANTECON,HECTOR" <sip:9492325109@jiveip.net>;tag=as6b8a3a35
t: <sip:0138387b18ea542af7000100620001@10.101.3.1:5061>
i: 4bfe07e05ec19c1a04e7c3675cec30a1@jiveip.net
CSeq: 103 INVITE
User-Agent: Jive Application Server
Authorization: Digest username="0138387b18ea542af7000100620001", realm="asterisk", algorithm=MD5, uri="sip:0138387b18ea542af7000100620001@10.101.3.1:5061", nonce="67552883", response="7794b7de9372617dd4be729db67dd2fa"
Date: Thu, 06 Feb 2014 22:14:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces
X-ref: 570615494_21262505@4.55.3.161
X-jid: pbx24.c1.jiveip.net-1391724857.2444415
Privacy: none
c: application/sdp
l: 320
P-hint: NAT
v=0
o=root 1782689202 1782689203 IN IP4 199.36.250.31
s=Jive
c=IN IP4 199.36.250.31
t=0 0
m=audio 24582 RTP/AVP 0 9 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
Here is my asterisk sip config:
[777]
deny=0.0.0.0/0.0.0.0
secret=tin03mar
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/777
mailbox=777@device
permit=0.0.0.0/0.0.0.0
callerid=Jeff Bennett <777>
callcounter=yes
faxdetect=no
[from-trunk]
host=199.36.250.0/24
type=friend
insecure=port,invite
dissallow=all
allow=ulaw
context=from-trunk
All suggestions are welcome
Thank you in advance.