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ds at caribenet.com
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 Posted: Tue Feb 26, 2008 12:31 pm Post subject: [asterisk-users] Sip trunk mystery |
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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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stotaro at totarotechn...
Guest
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| Posted: Tue Feb 26, 2008 12:43 pm Post subject: [asterisk-users] Sip trunk mystery |
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|
On Tue, Feb 26, 2008 at 12:31 PM, Dirk Enrique Seiffert
<ds at caribenet.com> wrote:
| Quote: | 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
|
Does it retransmit the invite six times and then hangup? When I have
seen this it was a firewall issue on the remote (provider) side.
Thanks,
Steve Totaro |
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ds at caribenet.com
Guest
|
| Posted: Tue Feb 26, 2008 2:51 pm Post subject: [asterisk-users] Sip trunk mystery |
|
|
Hi Steve,
| Quote: | Does it retransmit the invite six times and then hangup? When I have
seen this it was a firewall issue on the remote (provider) side.
|
Indeed it tries seven times. But I think this is the Asterisk default. The
same account configured in my Snom Phone works without problem, - from
same network to same network.
Thanks al lot
Enrique
Cartagena - Colombia
http://www.sipcolombia.com |
|
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jsmith at digium.com
Guest
|
| Posted: Tue Feb 26, 2008 3:21 pm Post subject: [asterisk-users] Sip trunk mystery |
|
|
On Tue, 2008-02-26 at 12:31 -0500, Dirk Enrique Seiffert wrote:
| Quote: | 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:
|
I'll take a stab at it.
| Quote: | First of all: Registry works!
|
Registering to another host doesn't mean anything when it comes to
sending them a call. Registration only tells them your IP address and
port so that they can send calls *to you*.
| Quote: | 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
---
|
Notice how the Contact Header and the SDP all have the IP address of
192.168.8.3? If your firewall isn't masquerading (rewriting) those
addresses as the SIP traffic goes through it, then the device on the
other end is going to try to contact 192.168.8.3, and I'm guessing it's
going to have a hard time doing that. (This would also explain why
you're seeing outbound traffic only in your tcpdump traces.)
--
Jared Smith
Community Relations Manager
Digium, Inc. |
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ds at caribenet.com
Guest
|
| Posted: Tue Feb 26, 2008 4:40 pm Post subject: [asterisk-users] Sip trunk mystery |
|
|
Hi Jared,
| Quote: |
Notice how the Contact Header and the SDP all have the IP address of
192.168.8.3? If your firewall isn't masquerading (rewriting) those
addresses as the SIP traffic goes through it, then the device on the
other end is going to try to contact 192.168.8.3, and I'm guessing it's
going to have a hard time doing that. (This would also explain why
you're seeing outbound traffic only in your tcpdump traces.)
|
My firewall is masquerading, anyhow I modified also the
externalip=mypublicIP . Now it looks like this:
Retransmitting #4 (NAT) to 190.144.151.212:5060:
INVITE sip:xxxxx46022 at sip.net-voz.com SIP/2.0
Via: SIP/2.0/UDP mypublicIP:5060;branch=z9hG4bK01757b08;rport
From: "901" <sip:901 at sip.net-voz.com>;tag=as69ce5a7a
To: <sip:5756646022 at sip.net-voz.com>
Contact: <sip:901 at mypublicIP>
Call-ID: 060bfbb24ecf6c88348d72d91e3ea978 at sip.net-voz.com
CSeq: 103 INVITE
User-Agent: Grandstream BT100 1.0.4.49
Max-Forwards: 70
Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com",
nonce="120406066426160405702174408208",
response="a6c72a47ea39c1200f2add823369ebce", opaque=""
Date: Tue, 26 Feb 2008 21:20:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
What else might might be wrong here?
Thanks
Enrique
Cartagena - Colombia
http://www.sipcolombia.com |
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