SOLVED - getting iptel working with freepbx - desperate - please help

General Help
1

Solved, the initial post below, solution:
peer details:
type=friend
secret=password
username=iptel-username
host=iptel.org
disallow=all
allow=ulaw
[email protected]
fromdomain=iptel.org
canreinvite=no
insecure=invite,port
qualify=yes
nat=no
context=from-trunk

then create inbound route and use “iptel-username” as DID

Hi all,

I have an iptel account and when I use a softphone like twinkle or x-lite i get the iptel account working without any problems. Now I also have a freepbx server and would like to have the iptel account in there as well. I have a working extension (called 900) to which i want to forward the incoming calls on iptel, somehow that does not seem to work.

I have setup a sip trunk for iptel with this info:

tunk-name: iptel
peer details:
type=friend
username=iptel-username
[email protected]
secret=mypassword
fromdomain=iptel.org
host=iptel.org

user-context: [email protected]
user details:
username=iptel-username
[email protected]
secret=mypassword
fromdomain=iptel.org
host=iptel.org
type=user
context=from-trunk

registry string:
[email protected]:[email protected]

when i do a “sip show registry” I get this:
Host dnsmgr Username Refresh State Reg.Time
iptel.org:5060 N iptel-username@ 225 Registered Sun, 16 Oct 2011 14:33:27

when i do a “sip show peers” I get this:
Name/username Host Dyn Forcerport ACL Port Status
iptel/iptel-username 213.192.59.75 5060 Unmonitored

however when calling my iptel, i get only a busy signal (though when registered in Twinkle or xlite the softphone rings and i get no busy signal, but normal ringing).

I also ran sip debug on:
SIP Debugging enabled

<— SIP read from UDP:213.192.59.75:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: sip:213.192.59.75;ftag=as2bdd6efb;avp=WL8DBwBhY2NvdW50AwB5ZXMDBgBzdGltZXIEADE4MDADCQBkaWFsb2dfaWQWADJkYzEtNGU4YzRiOGItMGViODNiYjQ;lr=on
Via: SIP/2.0/UDP 213.192.59.75;branch=z9hG4bKd9bf.d802a9d5.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK71508c13;rport=5060
From: “unknown” sip:[email protected];tag=as2bdd6efb
To: sip:[email protected]
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: IPKall
Max-Forwards: 16
Date: Sun, 16 Oct 2011 12:39:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 335
P-hint: usrloc applied
Session-Expires: 1800
X-RTP-Proxy: YES

v=0
o=root 30172 30172 IN IP4 66.54.140.46
s=session
c=IN IP4 213.192.59.91
t=0 0
m=audio 47950 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (19 headers 16 lines) —
Sending to 213.192.59.75:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘iptel’ for ‘2062040232’ from 213.192.59.75:5060

<— Reliably Transmitting (no NAT) to 213.192.59.75:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 213.192.59.75;branch=z9hG4bKd9bf.d802a9d5.0;received=213.192.59.75
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK71508c13;rport=5060
From: “unknown” sip:[email protected];tag=as2bdd6efb
To: sip:[email protected];tag=as509cae65
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="18661242"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:213.192.59.75:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 213.192.59.75;branch=z9hG4bKd9bf.d802a9d5.0
From: “unknown” sip:[email protected];tag=as2bdd6efb
To: sip:[email protected];tag=as509cae65
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 16
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.1.15:5060 —>
REGISTER sip:192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK-14814f81
From: line1 sip:[email protected];tag=2d21ed6ed03f7469o0
To: line1 sip:[email protected]
Call-ID: [email protected]
CSeq: 240550 REGISTER
Max-Forwards: 70
Authorization: Digest username=“900”,realm=“asterisk”,nonce=“36ba1657”,uri=“sip:192.168.1.102”,algorithm=MD5,response="8ca26ba736f8f818a0a5e9a9059f13c6"
Contact: line1 sip:[email protected]:5060;expires=60
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.15:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.15:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK-14814f81;received=192.168.1.15
From: line1 sip:[email protected];tag=2d21ed6ed03f7469o0
To: line1 sip:[email protected];tag=as53eb7fa7
Call-ID: [email protected]
CSeq: 240550 REGISTER
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0cf3e154"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.15:5060 —>
REGISTER sip:192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK-ed5031da
From: line1 sip:[email protected];tag=2d21ed6ed03f7469o0
To: line1 sip:[email protected]
Call-ID: [email protected]
CSeq: 240551 REGISTER
Max-Forwards: 70
Authorization: Digest username=“900”,realm=“asterisk”,nonce=“0cf3e154”,uri=“sip:192.168.1.102”,algorithm=MD5,response="8e1ddc99327d56c5e868745ecb199f28"
Contact: line1 sip:[email protected]:5060;expires=60
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.15:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.15:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK02a798ed
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4af859df
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Sun, 16 Oct 2011 12:39:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (no NAT) to 192.168.1.15:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK-ed5031da;received=192.168.1.15
From: line1 sip:[email protected];tag=2d21ed6ed03f7469o0
To: line1 sip:[email protected];tag=as53eb7fa7
Call-ID: [email protected]
CSeq: 240551 REGISTER
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: sip:[email protected]:5060;expires=60
Date: Sun, 16 Oct 2011 12:39:04 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.15:5060 —>
SIP/2.0 200 OK
To: sip:[email protected]:5060;tag=606ec192f35a0e95i0
From: “Unknown” sip:[email protected];tag=as4af859df
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK02a798ed
Server: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

pbx*CLI> sip set debug off
SIP Debugging Disabled

I have tried many different things, but have no idea how to continue anymore, please help me, thank you in advance,
Peter

2

I know this is an old thread, but it is one of the few I have found on FreePBX and iptel.

I am brand new to FreePBX and am having the same problem as described here. I tried creating a trunk with the “peer details:” listed above, and then an “inbound route”, but still get a “busy signal”. Has anyone else gotten this to work? If so, how?

If you need additional details, just ask and I’d be happy to provide them.

3

maybe make first sure that everything else works so to be clear that the problem is with the iptel config, meaning use another provider and set it up in the same way with an inbound route to an extension so you can be sure the problem is really the iptel config

have you created the inbound route and used your iptel username as DID entry?

4

Thanks for the reply. Sorry, I didn’t see it until just now.

have you created the inbound route and used your
iptel username as DID entry?

I believe so, but it’s entirely possible I did something wrong … I’m in the middle of something today, but I’d still love to figure this out. I’ll post back in the next day or so with the the exact settings I entered.

Thanks again for responding. Much appreciated!

5

I didn’t forget about this, but I’ve been busy with other tasks and haven’t had time to revisit this.