[asterisk-users] Sip trunk mystery
Dirk Enrique Seiffert
ds at caribenet.com
Tue Feb 26 11:31:20 CST 2008
Hello,
I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server.
The system is in production with local extensions, a zap trunk and a
working sip trunk with sipgate.de.
My asterisk server is behind a NAT/Firewall, anyhow it registers and works
well with sipgate.de on incoming and outgoing calls.
I aquired an account with a reseller net-voz.com: I did some testing with
the account directly from a Snom300 phone - works without a problem,
(behind the nat) I spent hours testing and adjusting the trunk
configuration for net-voz, maybe sombody on the list can give a helpful hint:
First of all: Registry works!
pbx*CLI> sip show registry
Host Username Refresh State
Reg.Time
sip.net-voz.com:5060 xxxxxx6168 585 Registered
Tue, 26 Feb 2008 10:47:58
sipgate.de:5060 xxxx0823 105 Registered
Tue, 26 Feb 2008 10:56:22
This is my config:
[ringtime]
username=5515816168
type=peer
type=friend
secret=118873
insecure=very
host=sip.net-voz.com
fromuser=5515816168
fromdomain=sip.net-voz.com
canreinvite=no
call-limit=50
I tried faking the user agent (without success)
useragent = Grandstream BT100 1.0.4.49
externip=xx.xx.116.229
localnet=192.168.8.0/255.255.255.0
On my gateway I can see the following with tcpdump:
listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes
11:05:57.386827 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
810
11:05:57.452414 IP 190.144.151.212.sip > pbx.lintec.sip: SIP, length: 442
11:05:57.453021 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length:
385
11:05:57.453587 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
11:05:58.452868 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
11:06:01.453814 IP pbx.mydomain.sip > 190.144.151.212.sip: SIP, length: 1030
On the astersik CLI the logs show:
Audio is at 192.168.8.3 port 14800
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 190.144.151.212:5060:
INVITE sip:5756646022 at sip.net-voz.com SIP/2.0
Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5
To: <sip:5756646022 at sip.net-voz.com>
Contact: <sip:5515816168 at 192.168.8.3>
Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com",
nonce="120404195526111105702055508208",
response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
Date: Tue, 26 Feb 2008 16:09:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 2381 2382 IN IP4 192.168.8.3
s=session
c=IN IP4 192.168.8.3
t=0 0
m=audio 14800 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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
---
Retransmitting #1 (no NAT) to 190.144.151.212:5060:
INVITE sip:5756646022 at sip.net-voz.com SIP/2.0
Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5
To: <sip:xxxxxx6022 at sip.net-voz.com>
Contact: <sip:xxxxx6168 at 192.168.8.3>
Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com",
nonce="120404195526111105702055508208",
response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque=""
Date: Tue, 26 Feb 2008 16:09:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
It looks like the comuunication starts but then gets lost.??
Any idea is appreciated.
Thanks
Enrique
Cartagena - Colombia
http://www.sipcolombia.com
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