VoIP Mailing List Archives

[noah at allresearch.com]

Hi,

I have our FS server configured and running.

I can register a polycom phone to it perfectly. Can make calls out
and receive calls in. Works well, quality is solid, etc.

Now, I want to register an asterisk box with FS. (Our eventual plan
is to resell VOIP as a "trunk" to end users, so this is an important
test.)

FS won't route the calls from asterisk. I don't see any specific
errors in the FS console, but the calls just die.

Below is the result of "sofia status profile default" showing that
both the asterisk box and polycom phone did successfully register.
(username, password, host changed for privacy)

Call-ID 456fa3406d67ee337c6c81264932f76a@127.0.0.1
User 3235551212@111.111.111.111
Contact "user" <sip: 3235551212@222.222.222.222:1024;fs_nat=yes>
Agent Asterisk PBX
Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 12:17:40)

Call-ID 8c5f3b02-c1f07a54-7d8ac0c7@10.0.1.110
User 3235551212@111.111.111.111
Contact "user" <sip: 3235551212@222.222.222.222:5060;fs_nat=yes>
Agent PolycomSoundPointIP-SPIP_500-UA/2.1.3.0028
Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 13:57:15)



Below is the config in my sip.conf for asterisk. (IP and DID changed
for privacy)

[Freeswitch]
host=111.111.111.111
username=3235551212
secret=password
port=5060
type=peer
trustrpid=yes
sendrpid=yes
context=from-trunk
canreinvite=no
disallow=all
allow=ulaw





Can anyone help me figure out what is wrong?


Thanks,

-N

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[brian at freeswitch.org]

you'll need to set from-domain in the sip.conf on asterisk

/b

On Sep 22, 2008, at 2:30 PM, Noah Silverman wrote:

[noah at allresearch.com]

Tried that and still doesn't work.

I've attached the SIP INVITE so that maybe you'll see something that
gives you a clue.

Also, I don't know if it matters, but the FS server is actually in an
off site data center. I'm connecting to it remotely from my office.
Works fine for a single polycom phone. (That, and the sound quality
is AMAZING! )

Here is what I have in Asterisk now...
[Freeswitch]
host=111.111.111.111
username=3235551212
secret=password
fromdomain=111.111.111.111
port=5060
type=peer
trustrpid=yes
sendrpid=yes
context=from-trunk
canreinvite=no
disallow=all
allow=ulaw



Here's the SIP INVITE. (IP's changed to protect the innocent.)
111.111.111.111 is the address of my FS server
222.222.222.222 is the address of my asterisk server
3235551212 is my username/did/account in FS

U 222.222.222.222:1024 -> 111.111.111.111:5060
INVITE sip:13237773456@111.111.111.111 SIP/2.0.
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport.
From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as146a87d7.
To: <sip:13237773456@111.111.111.111>.
Contact: <sip:3235551212@222.222.222.222>.
Call-ID: 4382446d3a46269d7f469e93147b46eb@111.111.111.111
CSeq: 103 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "3235551212" <sip:
3235551212@111.111.111.111>;privacy=off;screen=no.
Proxy-Authorization: Digest username="3235551212",
realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456@111.111.111.111
", nonce="c52ba984-f888-dd11-80f4-00188b37805b",
response="01635afacb7eeebc6fc0888991c0411d", qop=auth,
cnonce="4e065f27", nc=00000001.
Date: Mon, 22 Sep 2008 22:54:14 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 234.





On Sep 22, 2008, at 2:35 PM, Brian West wrote:

[jprangi at gmail.com]

Is your asterisk server behind the firewall or NAT.
Does your FS respond to the invite from asterisk,
Did you run sip trace on both asterisk and FS. I mean it will be useful to know if the below sip trace is from asterisk or FS. If from asterisk, then need to make sure if FS got the request and how it replied to that request.

Hope this will help you in debugging the problem.

Jai
www.didforsale.com
*Buy SIP DIDs all Over US at low cost, unlimited minutes
http://www.didforsale.com"

On Mon, Sep 22, 2008 at 4:01 PM, Noah Silverman <noah@allresearch.com (noah@allresearch.com)> wrote:

[noah at allresearch.com]

That SIP request was what was received on the FS server.


I just made two calls. One from the polylcom phone which worked and one through asterisk which failed. I'm comparing the two SIP INVITEs to see what the difference is. (Looking at the Invites that FS RECEIVED.)


Much is the same. here is what I found different.


To: On the successful call has "user=phone" appended. Nothing on the failed call


Call-ID: On the failed call is "abcdefetc:domain" On the successful call is "abcdetc:phone_ip" <--- Might be significant.


Remote-Party-ID: Exists for failed call. Does not exist for successful call


Proxy-Authorization: The uri for successful call has port 5060. The uri for failed call has no port <--- Might be significant.




Still digging through debug stuff but wanted to send this to the list incase someone had any ideas.


Thanks!!


-Noah



On Sep 22, 2008, at 4:36 PM, Jai Rangi wrote:

[brian at freeswitch.org]

What was the response and where did it go? I need to see the full sip trace along with console output.

TPORT_LOG=1 ./freeswitch


/b

On Sep 22, 2008, at 4:01 PM, Noah Silverman wrote:

[noah at allresearch.com]

NOTE: IP and DID changed.
Freeswitch server = 111.111.111.111
Asterisk server = 222.222.222.222
DID = 3235551212


---------------
recv 862 bytes from udp/[222.222.222.222]:1024 at 03:51:22.604274:
------------------------------------------------------------------------
INVITE [url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK38495771;rport
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>
Contact: <[url=sip:3235551212@10.0.1.100]sip:3235551212@10.0.1.100[/url]>
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;privacy=off;screen=no
Date: Tue, 23 Sep 2008 03:57:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 3463 3463 IN IP4 10.0.1.100
s=session
c=IN IP4 10.0.1.100
t=0 0
m=audio 13808 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------------------------------------------------------------------------
send 356 bytes to udp/[222.222.222.222]:1024 at 03:51:22.606561:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK38495771;rport=1024;received=222.222.222.222
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported
Content-Length: 0

------------------------------------------------------------------------
send 833 bytes to udp/[222.222.222.222]:1024 at 03:51:22.636580:
------------------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK38495771;rport=1024;received=222.222.222.222
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>;tag=yHjpHt1BmS7yS
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary
Proxy-Authenticate: Digest realm="111.111.111.111", nonce="bacd48e3-2289-dd11-80f4-00188b37805b", algorithm=MD5, qop="auth"
Content-Length: 0

------------------------------------------------------------------------
recv 481 bytes from udp/[222.222.222.222]:1024 at 03:51:22.669730:
------------------------------------------------------------------------
ACK [url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK38495771;rport
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>;tag=yHjpHt1BmS7yS
Contact: <[url=sip:3235551212@10.0.1.100]sip:3235551212@10.0.1.100[/url]>
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;privacy=off;screen=no
Content-Length: 0

------------------------------------------------------------------------
recv 1119 bytes from udp/[222.222.222.222]:1024 at 03:51:22.670427:
------------------------------------------------------------------------
INVITE [url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK59b179c5;rport
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>
Contact: <[url=sip:3235551212@10.0.1.100]sip:3235551212@10.0.1.100[/url]>
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;privacy=off;screen=no
Proxy-Authorization: Digest username="3235551212", realm="111.111.111.111", algorithm=MD5, uri="[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]", nonce="bacd48e3-2289-dd11-80f4-00188b37805b", response="ff3cdb8073becadfe55acfdaf5216441", qop=auth, cnonce="0fe2ecd1", nc=00000001
Date: Tue, 23 Sep 2008 03:57:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 3463 3464 IN IP4 10.0.1.100
s=session
c=IN IP4 10.0.1.100
t=0 0
m=audio 13808 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------------------------------------------------------------------------
send 356 bytes to udp/[222.222.222.222]:1024 at 03:51:22.673443:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK59b179c5;rport=1024;received=222.222.222.222
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 103 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported
Content-Length: 0

------------------------------------------------------------------------
send 691 bytes to udp/[222.222.222.222]:1024 at 03:51:22.784872:
------------------------------------------------------------------------
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK59b179c5;rport=1024;received=222.222.222.222
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>;tag=ZtBFKNjFH2XHN
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 103 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary
Content-Length: 0

------------------------------------------------------------------------
recv 481 bytes from udp/[222.222.222.222]:1024 at 03:51:22.811109:
------------------------------------------------------------------------
ACK [url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK59b179c5;rport
From: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;tag=as68ae65f8
To: <[url=sip:13237773456@111.111.111.111]sip:13237773456@111.111.111.111[/url]>;tag=ZtBFKNjFH2XHN
Contact: <[url=sip:3235551212@10.0.1.100]sip:3235551212@10.0.1.100[/url]>
Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 (1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111)
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "3235551212" <[url=sip:3235551212@111.111.111.111]sip:3235551212@111.111.111.111[/url]>;privacy=off;screen=no
Content-Length: 0

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






On Sep 22, 2008, at 7:00 PM, Brian West wrote: