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Fisher Networks
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Hey, I started our settings from scratch and appear to be having issues getting the ability to make outbound calls (and inbound, really). We use bandwidth.com SIP trunking and with their IP in the domain field, the same IP in the outbound gateway and our number as the DID in a SIP Gateway we can't get out. The only log I get is the following.

 

[5] 2008/02/04 17:52:21: Identify trunk (domain name match) 1

[5] 2008/02/04 17:52:21: Dialplan External: Match 206xxxxxxx@domain.com to <sip:2067690931@216.82.x.x;user=phone> on trunk BWGW1

 

It does not appear to be getting out at all at this point. The phone says this:

 

[5]4/2/2008 18:54:08: Dialog 7/6 going to trying

[5]4/2/2008 18:54:08: Dialog 7/6 going to early

[5]4/2/2008 18:54:08: Dialog 7/6 going to terminated

[5]4/2/2008 18:54:08: timeout::callback: Registering with timeout of 0 ms

[5]4/2/2008 18:54:08: timeout::callback: Registering with timeout of 0 ms

[2]4/2/2008 18:54:36: Registered at registrar as 501@domain.com

[0]4/2/2008 18:54:37: Webclient: Could not find host snom360.htm:80

[0]4/2/2008 18:54:37: Webclient: Could not find host snom360-00041323C1DB.htm:80

 

I think the [5]s are the only important logs there.

 

Anyway, the dial plan is a *, supposedly to accept any input and pass it along.

 

Any ideas what I am missing? I simplified the internal setup, so there are no hunt groups, attendants or anything but a single, registered extension.

 

Also, what ports do I need to make sure are open? I know of 5060 and 5061.

 

Thanks for your help!

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[5] 2008/02/04 17:52:21: Identify trunk (domain name match) 1

[5] 2008/02/04 17:52:21: Dialplan External: Match 206xxxxxxx@domain.com to <sip:2067690931@216.82.x.x;user=phone> on trunk BWGW1

 

I would set an outbound proxy on the trunk. Also, if you change the IP configuration, you should restart the service (just to be sure). If it does not help, turn SIP logging on and see what the PBX tries to send out.

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The outbound proxy should be the Trunking service IP or my gateway?

 

"SIP Logging" was on, but under that I enabled every type of SIP logging. However, the same (lack of) errors appear. However the SIP trace on the phone yields more info:

 

Sent to tls:10.0.1.3:5061 at 5/2/2008 01:00:28:050 (1254 bytes):

 

INVITE sip:206xxxxxxx@domain.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-dktogirkbrph;rport

From: "User" <sip:501@domain.com>;tag=9yc0cxo949

To: <sip:206xxxxxxx@domain.com;user=phone>

Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:501@10.0.1.109:4967;transport=tls;line=uexh66e7>;flow-id=1

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom360/6.2.3

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 471

 

v=0

o=root 1927523315 1927523315 IN IP4 10.0.1.109

s=call

c=IN IP4 10.0.1.109

t=0 0

m=audio 51080 RTP/AVP 0 8 9 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XUR1mT6ftFtPm4go7v37e/vtfmrfHzWmQ4OxofLW

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=encryption:optional

a=sendrecv

 

 

 

--------------------------------------------------------------------------------

 

Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:140 (323 bytes):

 

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-dktogirkbrph;rport=4967

From: "Tanya " <sip:501@domain.com>;tag=9yc0cxo949

To: <sip:206xxxxxxx@domain.com;user=phone>;tag=0e7ad1ccc9

Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB

CSeq: 1 INVITE

Content-Length: 0

 

 

 

 

--------------------------------------------------------------------------------

 

Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:150 (685 bytes):

 

NOTIFY sip:501@10.0.1.109:4967;transport=tls;line=uexh66e7 SIP/2.0

Via: SIP/2.0/TLS 10.0.1.3:5061;branch=z9hG4bK-0d5b74cb3e5a5bdc447bbbe02c21b327;rport

From: <sip:501@domain.com;user=phone>;tag=f7b6300e03

To: <sip:501@domain.com>;tag=oyj484iitx

Call-ID: 3c267009c832-wjx5tt2ysc3u@snom360-00041323C1DB

CSeq: 19600 NOTIFY

Max-Forwards: 70

Contact: <sip:10.0.1.3:5061;transport=tls>

Event: dialog

Subscription-State: active;expires=187

Content-Type: application/dialog-info+xml

Content-Length: 158

 

<?xml version="1.0"?>

<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="36" state="full" entity="sip:501@domain.com"></dialog-info>

 

<snip>

 

 

Sent to tls:10.0.1.3:5061 at 5/2/2008 01:00:28:380 (318 bytes):

 

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.1.3:5061;branch=z9hG4bK-24329e1bc69a2121a722c9dd45bdb3e5;rport=5061

From: <sip:501@domain.com;user=phone>;tag=f7b6300e03

To: <sip:501@domain.com>;tag=oyj484iitx

Call-ID: 3c267009c832-wjx5tt2ysc3u@snom360-00041323C1DB

CSeq: 19601 NOTIFY

Content-Length: 0

 

 

 

 

--------------------------------------------------------------------------------

 

Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:390 (402 bytes):

 

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-wweqrjre6ksm;rport=4967

From: "Tanya " <sip:501@domain.com>;tag=9yc0cxo949

To: <sip:206xxxxxxx@domain.com;user=phone>;tag=0e7ad1ccc9

Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB

CSeq: 2 PRACK

Contact: <sip:501@10.0.1.3:5061;transport=tls>

User-Agent: pbxnsip-PBX/2.1.5.2357

Content-Length: 0

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I set the outbound proxy to the server's internal NIC (that the clients all funnel through) and it said "Connection refused".

 

Set the outbound proxy of the trunk to your service providers address.

 

The outbound proxy on the phone should point to the PBX address. And BTW you better upgrade those phones to 7.1.30, otherwise you will experience problems with SRTP.

 

You better include the logs from the PBX, not from the phone - then it is easier to see the PBX perspective!

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Now we're getting somewhere. The logs now appear to be dial plan issues? I'm not too sure what the error is here besides the e164 message.

 

[7]2008/02/07 13:43:17: SIP Tx tls:10.0.1.109:3337:

SIP/2.0 404 Number not in e164 format, example +12125551212

Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-d67yw065242a;rport=3337

From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

CSeq: 1 INVITE

Contact: <sip:501@10.0.1.3:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.5.2357

Content-Length: 0

 

 

[7] 2008/02/07 13:43:17: SIP Rx tls:10.0.1.109:3337:

PRACK sip:501@10.0.1.3:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-8mic0lb2vaj9;rport

From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:501@10.0.1.109:3337;transport=tls;line=uexh66e7>;flow-id=1

RAck: 1 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer

Content-Length: 0

 

 

[7] 2008/02/07 13:43:17: SIP Tx tls:10.0.1.109:3337:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-8mic0lb2vaj9;rport=3337

From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

CSeq: 2 PRACK

Contact: <sip:501@10.0.1.3:5061;transport=tls>

User-Agent: pbxnsip-PBX/2.1.5.2357

Content-Length: 0

 

 

[7] 2008/02/07 13:43:18: SIP Rx tls:10.0.1.109:3337:

ACK sip:1206xxxxxxx@domain.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-d67yw065242a;rport

From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:501@10.0.1.109:3337;transport=tls;line=uexh66e7>;flow-id=1

Content-Length: 0

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