Incoming call from sipphone

Asterisk Asterisk Support
1

I am having trouble receiving calls from sipphone on asterisk. I have tried just about everything I can think of without success.

Any help would be appreciated.

Following is the trace I get when I try to call in. It looks like asterisk is rejecting the incoming call.

Asterisk 1.2.9.1 svn rev 34876, Copyright © 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘show license’ for details.

Connected to Asterisk 1.2.9.1 svn rev 34876 currently running on asterisk1 (pid = 2316)
asterisk1CLI>
Verbosity is at least 1
Core debug is at least 1
e[Kasterisk1CLI>
<-- SIP read from 192.168.1.114:5060:

— (0 headers 0 lines) Nat keepalive —

e[Kasterisk1*CLI>
Destroying call ‘2a10278c52a78c88175a108219676748@127.0.0.1’

e[Kasterisk1*CLI>
<-- SIP read from 198.65.166.131:5060:
INVITE sip:s@74.121.76.72 SIP/2.0
Record-Route: sip:198.65.166.131;ftag=3264e7229a7aa93d;lr
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK1ac.5975288.0
Via: SIP/2.0/UDP 192.168.1.112;rport=60155;received=74.121.76.72;branch=z9hG4bK3c614020cc1b2d38
From: “SipPhone” sip:17476970961@proxy01.sipphone.com;tag=3264e7229a7aa93d
To: sip:17476899521@proxy01.sipphone.com
Contact: sip:17476970961@74.121.76.72:60155;nat=yes
Supported: replaces
Call-ID: 7b46edfd71a52619@192.168.1.112
CSeq: 22987 INVITE
User-Agent: Grandstream BT110 1.0.8.23
Max-Forwards: 16
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 355
RemoteIP: 74.121.76.72
P-hint: usrloc routed
P-RTP-Proxy: YES (1)
P-Behind-NAT: Yes (1)

v=0
o=17476970961 8000 8000 IN IP4 192.168.1.112
s=SIP Call
c=IN IP4 198.65.166.131
t=0 0
m=audio 42994 RTP/AVP 0 8 4 18 2 99 9
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
a=nortpproxy:yes

— (19 headers 17 lines)—
Using INVITE request as basis request - 7b46edfd71a52619@192.168.1.112
Sending to 198.65.166.131 : 5060 (NAT)
Found peer 'Sipphone’
Reliably Transmitting (NAT) to 198.65.166.131:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK1ac.5975288.0;received=198.65.166.131
Via: SIP/2.0/UDP 192.168.1.112;rport=60155;received=74.121.76.72;branch=z9hG4bK3c614020cc1b2d38
From: “SipPhone” sip:17476970961@proxy01.sipphone.com;tag=3264e7229a7aa93d
To: sip:17476899521@proxy01.sipphone.com;tag=as42f9a389
Call-ID: 7b46edfd71a52619@192.168.1.112
CSeq: 22987 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:s@74.121.76.72
Proxy-Authenticate: Digest realm=“asterisk”, nonce="6c05ce61"
Content-Length: 0

Scheduling destruction of call ‘7b46edfd71a52619@192.168.1.112’ in 15000 ms

e[Kasterisk1*CLI>
<-- SIP read from 198.65.166.131:5060:
ACK sip:s@74.121.76.72 SIP/2.0
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK1ac.5975288.0
From: “SipPhone” sip:17476970961@proxy01.sipphone.com;tag=3264e7229a7aa93d
Call-ID: 7b46edfd71a52619@192.168.1.112
To: sip:17476899521@proxy01.sipphone.com;tag=as42f9a389
CSeq: 22987 ACK
Content-Length: 0

— (7 headers 0 lines)—

e[Kasterisk1*CLI>
<-- SIP read from 198.65.166.131:5060:
SIP/2.0 200 OK
Record-Route: sip:198.65.166.131;ftag=3264e7229a7aa93d;lr
Via: SIP/2.0/UDP 192.168.1.112;received=74.121.76.72;branch=z9hG4bK8c1ea600ea114cab
From: “SipPhone” sip:17476970961@proxy01.sipphone.com;tag=3264e7229a7aa93d
To: sip:17476899521@proxy01.sipphone.com;tag=000009E9155994DD
Call-ID: 7b46edfd71a52619@192.168.1.112
CSeq: 22988 BYE
Contact: sip:bye+novoicemail@198.65.166.130:5060;nat=yes
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
Warning: 392 198.65.166.130:5060 "Noisy feedback tells: pid=2513 req_src_ip=198.65.166.131 req_src_port=5060 in_uri=sip:announcement+novoicemail@198.65.166.130:5060 out_uri=sip:announcement+novoicemail@proxy01.sipphone.com:5060 via_cnt==0"
RemoteIP: 198.65.166.130

— (12 headers 0 lines)—

e[Kasterisk1*CLI>
<-- SIP read from 198.65.166.131:5060:
SIP/2.0 200 OK
Record-Route: sip:198.65.166.131;ftag=3264e7229a7aa93d;lr
Via: SIP/2.0/UDP 192.168.1.112;received=74.121.76.72;branch=z9hG4bK8c1ea600ea114cab
From: “SipPhone” sip:17476970961@proxy01.sipphone.com;tag=3264e7229a7aa93d
To: sip:17476899521@proxy01.sipphone.com;tag=000009E9155994DD
Call-ID: 7b46edfd71a52619@192.168.1.112
CSeq: 22988 BYE
Contact: sip:bye+novoicemail@198.65.166.130:5060;nat=yes
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
Warning: 392 198.65.166.130:5060 "Noisy feedback tells: pid=2513 req_src_ip=198.65.166.131 req_src_port=5060 in_uri=sip:announcement+novoicemail@198.65.166.130:5060 out_uri=sip:announcement+novoicemail@proxy01.sipphone.com:5060 via_cnt==0"
RemoteIP: 198.65.166.130

— (12 headers 0 lines)—

e[Kasterisk1*CLI>
<-- SIP read from 198.65.166.131:5060:
SIP/2.0 200 OK
Record-Route: sip:198.65.166.131;ftag=3264e7229a7aa93d;lr
Via: SIP/2.0/UDP 192.168.1.112;received=74.121.76.72;branch=z9hG4bK8c1ea600ea114cab
From: “SipPhone” sip:17476970961@proxy01.sipphone.com;tag=3264e7229a7aa93d
To: sip:17476899521@proxy01.sipphone.com;tag=000009E9155994DD
Call-ID: 7b46edfd71a52619@192.168.1.112
CSeq: 22988 BYE
Contact: sip:bye+novoicemail@198.65.166.130:5060;nat=yes
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
Warning: 392 198.65.166.130:5060 "Noisy feedback tells: pid=2513 req_src_ip=198.65.166.131 req_src_port=5060 in_uri=sip:announcement+novoicemail@198.65.166.130:5060 out_uri=sip:announcement+novoicemail@proxy01.sipphone.com:5060 via_cnt==0"
RemoteIP: 198.65.166.130

— (12 headers 0 lines)—

e[Kasterisk1*CLI>
<-- SIP read from 192.168.1.114:5060:

— (0 headers 0 lines) Nat keepalive —

e[Kasterisk1*CLI>
Destroying call ‘7b46edfd71a52619@192.168.1.112’

e[Kasterisk1CLI>
asterisk1CLI>
asterisk1*CLI>

2

also post sip.conf and relevant parts of extensions.conf?

3

sip.conf

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to

; solve translation problems.

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw

; If you need to answer unauthenticated calls, you should change this
; next line to ‘from-trunk’, rather than ‘from-sip-external’.
; You’ll know this is happening if when you call in you get a message
; saying “The number you have dialed is not in service. Please check the
; number and try again.”

;context = from-sip-external
; Send unknown SIP callers to this context

context = from-trunk ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

sip-nat.conf
externip=74.121.xx.xx
localnet=192.168.1.222
localmask=255.255.255.0
maxexpirey=180
defaultexpirey=160
tos=reliability
srvlookup=yes
nat=yes

sip-additional.conf

register=17476899521:xxxxxxxx@proxy01.sipphone.com

[FSipphone]
username=17476899521
type=user
secre=xxxxxxxxxx
qualify=no
insecure=very
host=proxy01.sipphone.com
fromuser=17476899521
context=from-trunk
canreinvite=no
nat=yes

[Sipphone]
username=17476899521
type=peer
secret=xxxxxxxxxx
host=proxy01.sipphone.com

4

I tried seting the verbose level to 9. I don’t get any activity when I try to call in on the SIP channel. Even though the trace showes asterisk receiving the invite as per the trace above.