[Asterisk-Users] Failed calls from SNOM100 to *

[Julien L.]

Hi all,

I've just been setting up such a phone with * and I'm facing some problems.

At this time there is no customized extensions & others, so I use extension
8500 (voicemail) in default config.

What I get when this extension is called from the SNOM is pasted below. The
phone ignores *'s invites and * sends a "BYE" to it. Please note the IP in
the "Contact" header... * also send on a port the phone doesn't seem to be
listening on.

* is running on 192.168.35.4, the phone is 192.168.35.110.

Thanks for any help.

===

Sip read: LI> 
INVITE sip:8500 at db.intra;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.21.108:5060;branch=z9hG4bK-2xxwsyff0kus
Max-Forwards: 70
From: <sip:snom100 at db.intra>;tag=yz77r1h8iq
To: <sip:8500 at db.intra;user=phone>
Call-ID: 3eb464dda940-lvrq6eojtnl9 at 192.168.35.110
CSeq: 1 INVITE
Contact: <sip:snom100 at 192.168.35.110:5060>
User-Agent: snom Version 1.15u
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE
Supported: timer, 100rel, replaces
Session-Expires: 7200
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 6202 6202 IN IP4 192.168.35.110
s=SIP Call
c=IN IP4 192.168.35.110
t=0 0
m=audio 10016 RTP/AVP 18 3 0 8 101
a=rtpmap:18 g729/8000
a=rtpmap:3 gsm/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

16 headers, 12 lines
Using latest request as basis request
Sending to 192.168.35.110 : 5060 (non-NAT)
Capabilities: us - 2147483647, them - 270, combined - 270
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 8500 in default
list_route: hop: <sip:snom100 at 192.168.35.110:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.35.110:5060;branch=z9hG4bK-2xxwsyff0kus
From: <sip:snom100 at db.intra>;tag=yz77r1h8iq
To: <sip:8500 at db.intra;user=phone>;tag=as7f56f49b
Call-ID: 3eb464dda940-lvrq6eojtnl9 at 192.168.35.110
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:8500 at 0.0.19.196>
Content-Length: 0


 to 192.168.35.110:5060
We're at 0.0.19.196 port 39584
Answering with preferred capability 2147483647
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.35.110:5060;branch=z9hG4bK-2xxwsyff0kus
From: <sip:snom100 at db.intra>;tag=yz77r1h8iq
To: <sip:8500 at db.intra;user=phone>;tag=as7f56f49b
Call-ID: 3eb464dda940-lvrq6eojtnl9 at 192.168.35.110
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:8500 at 0.0.19.196>
Content-Type: application/sdp
Content-Length: 103

v=0
o=root 1455 1455 IN IP4 0.0.19.196
s=session
c=IN IP4 0.0.19.196
t=0 0
m=audio 39584 RTP/AVP


Trace during call init:

02:59:22.733391 192.168.35.4.5060 > 192.168.35.110.5060: udp 462 (DF) (ttl
64, id 5118, len 490)
02:59:22.734266 192.168.35.4.10956 > 192.168.35.110.10018: [udp sum ok] udp
45 (DF) [tos 0x10]  (ttl 64, id 22749, len 73)
02:59:22.753368 192.168.35.4.10956 > 192.168.35.110.10018: [udp sum ok] udp
45 (DF) [tos 0x10]  (ttl 64, id 25365, len 73)
[...]
02:59:22.873909 192.168.35.110 > 192.168.35.4: icmp: 192.168.35.110 udp
port 10018 unreachable for 192.168.35.4.10956 > 192.168.35.110.10018: udp
45 (DF) [tos 0x10]  (ttl 64, id 14636, len 73) [tos 0xd0]  (ttl 255, id
43493, len 101)
[...]


-- 
Julien Lesaint.