Outgoing call -> sometimes service unavailable on Cisco phon

Asterisk Asterisk Support
1

Hello,

We have setup an asterisk pabx (taridium) together with 2 weepee SIP accounts.
Sometimes when we make an outgoing call on our Cisco SP502G phones we get service unavailable when dialing a number.
I started debugging through the CentOS console where asterisk is running on and set the sip debug on.
The result is this:

Retransmitting #1 (NAT) to 91.208.12.132:5060:
INVITE sip:056782311@ssw2.weepee.org SIP/2.0
v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK135dcd82;rport
Max-Forwards: 70
f: “receptie” sip:056962000@81.82.255.37;tag=as52a904a0
t: sip:056782311@ssw2.weepee.org
m: sip:056962000@81.82.255.37:5060
i: 22970773392bef867fe3c36c6a6d6e40@81.82.255.37:5060
CSeq: 103 INVITE
User-Agent: taridium ipbx
Authorization: Digest username=“329909014559”, realm=“weepee”, algorithm=MD5, uri="sip:056782311@ssw2.weepee.org", nonce=“5c1fe103”, response="c42960de6430343333562e8384b6916d"
Date: Thu, 23 Aug 2012 09:01:19 GMT
x: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
c: application/sdp
l: 253

v=0
o=root 1265841916 1265841917 IN IP4 81.82.255.37
s=taridium ipbx
c=IN IP4 81.82.255.37
t=0 0
m=audio 18388 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=ptime:20
a=sendrecv

<— SIP read from UDP:91.208.12.132:5060 —>
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK135dcd82;received=81.82.255.37;rport=5060
From: “receptie” sip:056962000@81.82.255.37;tag=as52a904a0
To: sip:056782311@ssw2.weepee.org;tag=as75e8a8b7
Call-ID: 22970773392bef867fe3c36c6a6d6e40@81.82.255.37:5060
CSeq: 103 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 91.208.12.132:5060:
ACK sip:056782311@ssw2.weepee.org SIP/2.0
v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK135dcd82;rport
Max-Forwards: 70
f: “receptie” sip:056962000@81.82.255.37;tag=as52a904a0
t: sip:056782311@ssw2.weepee.org;tag=as75e8a8b7
m: sip:056962000@81.82.255.37:5060
i: 22970773392bef867fe3c36c6a6d6e40@81.82.255.37:5060
CSeq: 103 ACK
User-Agent: taridium ipbx
l: 0

–
[Aug 23 11:01:20] WARNING[28470]: chan_sip.c:20437 handle_response_invite: Received response: “Forbidden” from '“receptie” sip:056962000@81.82.255.37;tag=as52a904a0’
Scheduling destruction of SIP dialog ‘22970773392bef867fe3c36c6a6d6e40@81.82.255.37:5060’ in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– <SIP/11-000000dc>AGI Script ipbx/inside.ipbx completed, returning 0
– Auto fallthrough, channel ‘SIP/11-000000dc’ status is ‘CHANUNAVAIL’

<— Reliably Transmitting (no NAT) to 192.168.121.128:5060 —>
SIP/2.0 503 Service Unavailable
v: SIP/2.0/UDP 192.168.121.128:5060;branch=z9hG4bK-9d25c1ba;received=192.168.121.128
f: “receptie” sip:11@192.168.121.252;tag=9809033d15bc7b02o0
t: sip:056782311@192.168.121.252;tag=as284aaf35
i: 98d03819-35f56bd6@192.168.121.128
CSeq: 102 INVITE
Server: taridium ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
l: 0

<------------>
– Executing [h@inside:1] Hangup(“SIP/11-000000dc”, “”) in new stack
== Spawn extension (inside, h, 1) exited non-zero on ‘SIP/11-000000dc’

<— SIP read from UDP:192.168.121.128:5060 —>
ACK sip:056782311@192.168.121.252 SIP/2.0
Via: SIP/2.0/UDP 192.168.121.128:5060;branch=z9hG4bK-9d25c1ba
From: “receptie” sip:11@192.168.121.252;tag=9809033d15bc7b02o0
To: sip:056782311@192.168.121.252;tag=as284aaf35
Call-ID: 98d03819-35f56bd6@192.168.121.128
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“11”,realm=“ipbx”,nonce=“4c4247b4”,uri="sip:056782311@192.168.121.252",algorithm=MD5,response="644bf3ebcf7f4564e1b813b14cbe4b9a"
Contact: “receptie” sip:11@192.168.121.128:5060
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘98d03819-35f56bd6@192.168.121.128’ Method: ACK

<— SIP read from UDP:91.208.12.139:5060 —>
OPTIONS sip:s@81.82.255.37:5060 SIP/2.0
Via: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK696957e3;rport
Max-Forwards: 70
From: “weepee” sip:weepee@91.208.12.139;tag=as195efae5
To: sip:s@81.82.255.37:5060
Contact: sip:weepee@91.208.12.139:5060
Call-ID: 2886e2e2490f8cdb0469a7884a10ad93@91.208.12.139:5060
CSeq: 102 OPTIONS
User-Agent: weepee
Date: Thu, 23 Aug 2012 09:01:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.139:5060 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK696957e3;rport;received=91.208.12.139
f: “weepee” sip:weepee@91.208.12.139;tag=as195efae5
t: sip:s@81.82.255.37:5060;tag=as77dd1190
i: 2886e2e2490f8cdb0469a7884a10ad93@91.208.12.139:5060
CSeq: 102 OPTIONS
Server: taridium ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
m: sip:81.82.255.37:5060
Accept: application/sdp
l: 0

Does somebody know what to do? It is not clear if the problem is for Taridium (pabx) or for Weepee (sip provider).
Now we can not always make outgoing calls… Sometimes after 2 minutes is working again.

Via the webinterface from taridium, we created 2 sip trunks (account X and account Y) and 2 sip registrations (account X and acount Y). Each have there own servers at Weepee. Both accounts are in the PABX, the PABX is connected on our LAN and is using only 1 WAN IP address to the internet.
It looks that with 1 account the problem is less but it still there.

2

You’ve had a retransmission; you should have taken the log back to the start of the INVITE, for which there would typically be a 401 response, and included the first attempt of the transmission that timed out.

You have then be rejected for providing an invalid user/password.

My guess is that the downstream system has broken handling of retransmitted requests.

3

Hi,

We’re waiting for our provider to debug on their side. But it’ll be for later on this week or next week.
So in meanwhile the full log.

The problem occurs when we add a second SIP trunk. With just one SIP trunk it’s much more stable but however every now and then the “service unavailable” problem still occurs. But we can work with that for now. With 2 SIP trunks it’s impossible.

[quote]
<------------->
— (13 headers 0 lines) —
Looking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.139:5060 —>
SIP/2.0 200 OK

v: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK650ddd0c;rport;received=91.208.12.139

f: “weepee” sip:weepee@91.208.12.139;tag=as54750053

t: sip:s@81.82.255.37:5060;tag=as00688720

i: 11500e142cd592ec5862446a5d6eb76a@91.208.12.139:5060

CSeq: 102 OPTIONS

Server: taridium ipbx

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

k: replaces, timer

m: sip:81.82.255.37:5060

Accept: application/sdp

l: 0

<------------>
Scheduling destruction of SIP dialog ‘11500e142cd592ec5862446a5d6eb76a@91.208.12.139:5060’ in 32000 ms (Method: OPTIONS)

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.139:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK50f353d8;received=81.82.255.37;rport=5060

From: sip:329909014559@ssw9.brussels.weepee.org;tag=as7f32cec7

To: sip:329909014559@ssw9.brussels.weepee.org;tag=as1a822111

Call-ID: 01c6fd81421b23f77047aacd4ddaea40@127.0.0.1

CSeq: 145 REGISTER

Server: weepee

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

Supported: replaces, timer

Expires: 60

Contact: sip:s@81.82.255.37:5060;expires=60

Date: Mon, 27 Aug 2012 08:00:16 GMT

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘01c6fd81421b23f77047aacd4ddaea40@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Aug 27 09:47:23] e[1;33mNOTICEe[0m[25345]: e[1;37mchan_sip.ce[0m:e[1;37m20905e[0m e[1;37mhandle_response_registere[0m: Outbound Registration: Expiry for ssw9.brussels.weepee.org is 60 sec (Scheduling reregistration in 45 s)

e[Kipbx*CLI>
e[0KReally destroying SIP dialog ‘3c8cf6cb6c4326772a6823762428eeaf@91.208.12.131:5060’ Method: OPTIONS

e[Kipbx*CLI>
e[0KReally destroying SIP dialog ‘3a8bd7671f2367ae254b11c62af88728@91.208.12.139:5060’ Method: OPTIONS

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.131:5060 —>
OPTIONS sip:s@81.82.255.37:5060 SIP/2.0

Via: SIP/2.0/UDP 91.208.12.131:5060;branch=z9hG4bK22e80987;rport

Max-Forwards: 70

From: “weepee” sip:weepee@91.208.12.131;tag=as4a806ba1

To: sip:s@81.82.255.37:5060

Contact: sip:weepee@91.208.12.131:5060

Call-ID: 6721901b2bb3b8237f45a21b5c324490@91.208.12.131:5060

CSeq: 102 OPTIONS

User-Agent: weepee

Date: Mon, 27 Aug 2012 08:00:41 GMT

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

Supported: replaces, timer

Content-Length: 0

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

e[Kipbx*CLI>
e[0KLooking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.131:5060 —>
SIP/2.0 200 OK

v: SIP/2.0/UDP 91.208.12.131:5060;branch=z9hG4bK22e80987;rport;received=91.208.12.131

f: “weepee” sip:weepee@91.208.12.131;tag=as4a806ba1

t: sip:s@81.82.255.37:5060;tag=as3e68814d

i: 6721901b2bb3b8237f45a21b5c324490@91.208.12.131:5060

CSeq: 102 OPTIONS

Server: taridium ipbx

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

k: replaces, timer

m: sip:81.82.255.37:5060

Accept: application/sdp

l: 0

<------------>
Scheduling destruction of SIP dialog ‘6721901b2bb3b8237f45a21b5c324490@91.208.12.131:5060’ in 32000 ms (Method: OPTIONS)

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.139:5060 —>
OPTIONS sip:s@81.82.255.37:5060 SIP/2.0

Via: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK12f3657e;rport

Max-Forwards: 70

From: “weepee” sip:weepee@91.208.12.139;tag=as187ee071

To: sip:s@81.82.255.37:5060

Contact: sip:weepee@91.208.12.139:5060

Call-ID: 24848ca90919deb44622225834b7c449@91.208.12.139:5060

CSeq: 102 OPTIONS

User-Agent: weepee

Date: Mon, 27 Aug 2012 08:00:41 GMT

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

Supported: replaces, timer

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.139:5060 —>
SIP/2.0 200 OK

v: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK12f3657e;rport;received=91.208.12.139

f: “weepee” sip:weepee@91.208.12.139;tag=as187ee071

t: sip:s@81.82.255.37:5060;tag=as69a29d8a

i: 24848ca90919deb44622225834b7c449@91.208.12.139:5060

CSeq: 102 OPTIONS

Server: taridium ipbx

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

k: replaces, timer

m: sip:81.82.255.37:5060

Accept: application/sdp

l: 0

<------------>
Scheduling destruction of SIP dialog ‘24848ca90919deb44622225834b7c449@91.208.12.139:5060’ in 32000 ms (Method: OPTIONS)

e[Kipbx*CLI>
e[0KReally destroying SIP dialog ‘4ff391486e28cc427b09af591e087997@91.208.12.131:5060’ Method: OPTIONS

e[Kipbx*CLI>
e[0KReally destroying SIP dialog ‘3d971e8603d1ee695b04aa4933fa1e53@127.0.0.1’ Method: REGISTER

e[Kipbx*CLI>
e[0KReally destroying SIP dialog ‘11500e142cd592ec5862446a5d6eb76a@91.208.12.139:5060’ Method: OPTIONS

e[Kipbx*CLI>
e[0KReally destroying SIP dialog ‘01c6fd81421b23f77047aacd4ddaea40@127.0.0.1’ Method: REGISTER

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:192.168.121.114:5060 —>
INVITE sip:070222466@192.168.121.252 SIP/2.0

Via: SIP/2.0/UDP 192.168.121.114:5060;branch=z9hG4bK-6500258f

From: “stijn” sip:12@192.168.121.252;tag=1864b45a29a00c57o0

To: sip:070222466@192.168.121.252

Call-ID: c1729ad6-3d98f80b@192.168.121.114

CSeq: 101 INVITE

Max-Forwards: 70

Contact: “stijn” sip:12@192.168.121.114:5060

Expires: 240

User-Agent: Cisco/SPA502G-7.4.8a

Content-Length: 405

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE

Supported: replaces

Content-Type: application/sdp

v=0

o=- 33854058 33854058 IN IP4 192.168.121.114

s=-

c=IN IP4 192.168.121.114

t=0 0

m=audio 16396 RTP/AVP 0 2 8 9 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

<------------->
— (14 headers 18 lines) —
Sending to 192.168.121.114:5060 (no NAT)
Using INVITE request as basis request - c1729ad6-3d98f80b@192.168.121.114
Found peer ‘12’ for ‘12’ from 192.168.121.114:5060

<— Reliably Transmitting (no NAT) to 192.168.121.114:5060 —>
SIP/2.0 401 Unauthorized

v: SIP/2.0/UDP 192.168.121.114:5060;branch=z9hG4bK-6500258f;received=192.168.121.114

f: “stijn” sip:12@192.168.121.252;tag=1864b45a29a00c57o0

t: sip:070222466@192.168.121.252;tag=as12f2296f

i: c1729ad6-3d98f80b@192.168.121.114

C
e[Kipbx*CLI>
e[0KSeq: 101 INVITE

Server: taridium ipbx

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

k: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“ipbx”, nonce=“0bc1e52d”

l: 0

<------------>
Scheduling destruction of SIP dialog ‘c1729ad6-3d98f80b@192.168.121.114’ in 32000 ms (Method: INVITE)

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:192.168.121.114:5060 —>
ACK sip:070222466@192.168.121.252 SIP/2.0

Via: SIP/2.0/UDP 192.168.121.114:5060;branch=z9hG4bK-6500258f

From: “stijn” sip:12@192.168.121.252;tag=1864b45a29a00c57o0

To: sip:070222466@192.168.121.252;tag=as12f2296f

Call-ID: c1729ad6-3d98f80b@192.168.121.114

CSeq: 101 ACK

Max-Forwards: 70

Contact: “stijn” sip:12@192.168.121.114:5060

User-Agent: Cisco/SPA502G-7.4.8a

Content-Length: 0

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

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:192.168.121.114:5060 —>
INVITE sip:070222466@192.168.121.252 SIP/2.0

Via: SIP/2.0/UDP 192.168.121.114:5060;branch=z9hG4bK-384af06f

From: “stijn” sip:12@192.168.121.252;tag=1864b45a29a00c57o0

To: sip:070222466@192.168.121.252

Call-ID: c1729ad6-3d98f80b@192.168.121.114

CSeq: 102 INVITE

Max-Forwards: 70

Authorization: Digest username=“12”,realm=“ipbx”,nonce=“0bc1e52d”,uri="sip:070222466@192.168.121.252",algorithm=MD5,response=“e2310465256fdec778a4261db6e7ef41”

Contact: “stijn” sip:12@192.168.121.114:5060

Expires: 240

User-Agent: Cisco/SPA502G-7.4.8a

Content-Length: 405

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE

Supported: replaces

Content-Type: application/sdp

v=0

o=- 33854058 33854058 IN IP4 192.168.121.114

s=-

c=IN IP4 192.168.121.114

t=0 0

m=audio 16396 RTP/AVP 0 2 8 9 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

<------------->
— (15 headers 18 lines) —
Sending to 192.168.121.114:5060 (no NAT)
Using INVITE request as basis request - c1729ad6-3d98f80b@192.168.121.114
Found peer ‘12’ for ‘12’ from 192.168.121.114:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 9

e[Kipbx*CLI>
e[0KFound RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x111c (ulaw|alaw|g729|g726aal2|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.121.114:16396
Looking for 070222466 in inside (domain 192.168.121.252)
list_route: hop: sip:12@192.168.121.114:5060

<— Transmitting (no NAT) to 192.168.121.114:5060 —>
SIP/2.0 100 Trying

v: SIP/2.0/UDP 192.168.121.114:5060;branch=z9hG4bK-384af06f;received=192.168.121.114

f: “stijn” sip:12@192.168.121.252;tag=1864b45a29a00c57o0

t: sip:070222466@192.168.121.252

i: c1729ad6-3d98f80b@192.168.121.114

CSeq: 102 INVITE

Server: taridium ipbx

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

k: replaces, timer

m: sip:070222466@192.168.121.252:5060

l: 0

<------------>

e[Kipbx*CLI>
e[0K – Executing [070222466@inside:1] e[1;36mAGIe[0m(“e[1;35mSIP/12-000000dee[0m”, “e[1;35mipbx/inside.ipbxe[0m”) in new stack

e[Kipbx*CLI>
e[0K – Launched AGI Script /var/lib/asterisk/agi-bin/ipbx/inside.ipbx

e[Kipbx*CLI>
e[0K – AGI Script Executing Application: (Set) Options: (CALLERID(num)=056962000)

e[Kipbx*CLI>
e[0K – AGI Script Executing Application: (Dial) Options: (SIP/weepee_anylink/070222466)

e[Kipbx*CLI>
e[0K == Using SIP RTP TOS bits 184

e[Kipbx*CLI>
e[0K == Using SIP RTP CoS mark 5

e[Kipbx*CLI>
e[0KAudio is at 19610

e[Kipbx*CLI>
e[0KAdding codec 0x4 (ulaw) to SDP

e[Kipbx*CLI>
e[0KAdding codec 0x8 (alaw) to SDP

e[Kipbx*CLI>
e[0KAdding non-codec 0x1 (telephone-event) to SDP

e[Kipbx*CLI>
e[0KReliably Transmitting (NAT) to 91.208.12.132:5060:
INVITE sip:070222466@ssw2.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK36d8639c;rport

Max-Forwards: 70

f: “stijn” sip:056962000@81.82.255.37;tag=as74e569cc

t: sip:070222466@ssw2.weepee.org

m: sip:056962000@81.82.255.37:5060

i: 2746cb072a9d188443dec790070432aa@81.82.255.37:5060

CSeq: 102 INVITE

User-Agent: taridium ipbx

Date: Mon, 27 Aug 2012 07:47:57 GMT

x: 1800

Min-SE: 90

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

k: replaces, timer

c: application/sdp

l: 249

v=0

o=root 92987535 92987535 IN IP4 81.82.255.37

s=taridium ipbx

c=IN IP4 81.82.255.37

t=0 0

m=audio 19610 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=ptime:20

a=sendrecv

e[Kipbx*CLI>
e[0K – Called SIP/weepee_anylink/070222466

e[Kipbx*CLI>
e[0KRetransmitting #1 (NAT) to 91.208.12.132:5060:
INVITE sip:070222466@ssw2.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK36d8639c;rport

Max-Forwards: 70

f: “stijn” sip:056962000@81.82.255.37;tag=as74e569cc

t: sip:070222466@ssw2.weepee.org

m: sip:056962000@81.82.255.37:5060

i: 2746cb072a9d188443dec790070432aa@81.82.255.37:5060

CSeq: 102 INVITE

User-Agent: taridium ipbx

Date: Mon, 27 Aug 2012 07:47:57 GMT

x: 1800

Min-SE: 90

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

k: replaces, timer

c: application/sdp

l: 249

v=0

o=root 92987535 92987535 IN IP4 81.82.255.37

s=taridium ipbx

c=IN IP4 81.82.255.37

t=0 0

m=audio 19610 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=ptime:20

a=sendrecv

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.132:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK36d8639c;received=81.82.255.37;rport=5060

From: “stijn” sip:056962000@81.82.255.37;tag=as74e569cc

To: sip:070222466@ssw2.weepee.org;tag=as4fafefbc

Call-ID: 2746cb072a9d188443dec790070432aa@81.82.255.37:5060

CSeq: 102 INVITE

Server: weepee

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“weepee”, nonce=“060398b1”

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 91.208.12.132:5060:
ACK sip:070222466@ssw2.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK36d8639c;rport

Max-Forwards: 70

f: “stijn” sip:056962000@81.82.255.37;tag=as74e569cc

t: sip:070222466@ssw2.weepee.org;tag=as4fafefbc

m: sip:056962000@81.82.255.37:5060

i: 2746cb072a9d188443dec790070432aa@81.82.255.37:5060

CSeq: 102 ACK

User-Agent: taridium ipbx

l: 0

Audio is at 19610
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.208.12.132:5060:
INVITE sip:070222466@ssw2.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK3d468e50;rport

Max-Forwards: 70

f: “stijn” sip:056962000@81.82.255.37;tag=as74e569cc

t: sip:070222466@ssw2.weepee.org

m: sip:056962000@81.82.255.37:5060

i: 2746cb072a9d188443dec790070432aa@81.82.255.37:5060

CSeq: 103 INVITE

User-Agent: taridium ipbx

Authorization: Digest username=“329909014559”, realm=“weepee”, algorithm=MD5, uri="sip:070222466@ssw2.weepee.org", nonce=“060398b1”, response=“31a6c5665b88d1ad354e286b4fbb40df”

Date: Mon, 27 Aug 2012 07:47:58 GMT

x: 1800

Min-SE: 90

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

k: replaces, timer

c: application/sdp

l: 249

v=0

o=root 92987535 92987536 IN IP4 81.82.255.37

s=taridium ipbx

c=IN IP4 81.82.255.37

t=0 0

m=audio 19610 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=ptime:20

a=sendrecv

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.132:5060 —>
SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK3d468e50;received=81.82.255.37;rport=5060

From: “stijn” sip:056962000@81.82.255.37;tag=as74e569cc

To: sip:070222466@ssw2.weepee.org;tag=as4fafefbc

Call-ID: 2746cb072a9d188443dec790070432aa@81.82.255.37:5060

CSeq: 103 INVITE

Server: weepee

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

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 91.208.12.132:5060:
ACK sip:070222466@ssw2.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK3d468e50;rport

Max-Forwards: 70

f: “stijn” sip:056962000@81.82.255.37;tag=as74e569cc

t: sip:070222466@ssw2.weepee.org;tag=as4fafefbc

m: sip:056962000@81.82.255.37:5060

i: 2746cb072a9d188443dec790070432aa@81.82.255.37:5060

CSeq: 103 ACK

User-Agent: taridium ipbx

l: 0

[Aug 27 09:47:59] e[1;31mWARNINGe[0m[25345]: e[1;37mchan_sip.ce[0m:e[1;37m20437e[0m e[1;37mhandle_response_invitee[0m: Received response: “Forbidden” from ‘“stijn” sip:056962000@81.82.255.37;tag=as74e569cc’

e[Kipbx*CLI>
e[0KScheduling destruction of SIP dialog ‘2746cb072a9d188443dec790070432aa@81.82.255.37:5060’ in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)

e[Kipbx*CLI>
e[0K – <SIP/12-000000de>AGI Script ipbx/inside.ipbx completed, returning 0

e[Kipbx*CLI>
e[0K – Auto fallthrough, channel ‘SIP/12-000000de’ status is ‘CHANUNAVAIL’

<— Reliably Transmitting (no NAT) to 192.168.121.114:5060 —>
SIP/2.0 503 Service Unavailable

v: SIP/2.0/UDP 192.168.121.114:5060;branch=z9hG4bK-384af06f;received=192.168.121.114

f: “stijn” sip:12@192.168.121.252;tag=1864b45a29a00c57o0

t: sip:070222466@192.168.121.252;tag=as2e5c875d

i: c1729ad6-3d98f80b@192.168.121.114

CSeq: 102 INVITE

Server: taridium ipbx

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

k: replaces, timer

X-Asterisk-HangupCause: Call Rejected

X-Asterisk-HangupCauseCode: 21

l: 0

<------------>
– Executing [h@inside:1] e[1;36mHangupe[0m(“e[1;35mSIP/12-000000dee[0m”, “e[1;35me[0m”) in new stack
== Spawn extension (inside, h, 1) exited non-zero on ‘SIP/12-000000de’

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:192.168.121.114:5060 —>
ACK sip:070222466@192.168.121.252 SIP/2.0

Via: SIP/2.0/UDP 192.168.121.114:5060;branch=z9hG4bK-384af06f

From: “stijn” sip:12@192.168.121.252;tag=1864b45a29a00c57o0

To: sip:070222466@192.168.121.252;tag=as2e5c875d

Call-ID: c1729ad6-3d98f80b@192.168.121.114

CSeq: 102 ACK

Max-Forwards: 70

Authorization: Digest username=“12”,realm=“ipbx”,nonce=“0bc1e52d”,uri="sip:070222466@192.168.121.252",algorithm=MD5,response=“e2310465256fdec778a4261db6e7ef41”

C
e[Kipbx*CLI>
e[0Kontact: “stijn” sip:12@192.168.121.114:5060

User-Agent: Cisco/SPA502G-7.4.8a

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘c1729ad6-3d98f80b@192.168.121.114’ Method: ACK

e[Kipbx*CLI>
e[0K[Aug 27 09:48:07] e[1;33mNOTICEe[0m[25345]: e[1;37mchan_sip.ce[0m:e[1;37m13151e[0m e[1;37msip_reregistere[0m: – Re-registration for 329909014591@ssw1.brussels.weepee.org

e[Kipbx*CLI>
e[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.208.12.131:5060:
REGISTER sip:ssw1.brussels.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK126c8a7c

Max-Forwards: 70

f: sip:329909014591@ssw1.brussels.weepee.org;tag=as22241ac1

t: sip:329909014591@ssw1.brussels.weepee.org

i: 3d971e8603d1ee695b04aa4933fa1e53@127.0.0.1

CSeq: 146 REGISTER

User-Agent: taridium ipbx

Authorization: Digest username=“329909014591”, realm=“weepee”, algorithm=MD5, uri=“sip:ssw1.brussels.weepee.org”, nonce=“2a93cd7c”, response=“65aa4ffdd2f4f6afd713c13b6f8a33a1”

Expires: 30

m: sip:s@81.82.255.37:5060

l: 0

<— SIP read from UDP:91.208.12.131:5060 —>
OPTIONS sip:s@81.82.255.37:5060 SIP/2.0

Via: SIP/2.0/UDP 91.208.12.131:5060;branch=z9hG4bK1f75b031;rport

Max-Forwards: 70

From: “weepee” sip:weepee@91.208.12.131;tag=as3ef22ef5

To: sip:s@81.82.255.37:5060

Contact: sip:weepee@91.208.12.131:5060

Call-ID: 06e6700b7dc1632c037f396d59853719@91.208.12.131:5060

CSeq: 102 OPTIONS

User-Agent: weepee

Date: Mon, 27 Aug 2012 08:01:06 GMT

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

Supported: replaces, timer

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.131:5060 —>
SIP/2.0 200 OK

v: SIP/2.0/UDP 91.208.12.131:5060;branch=z9hG4bK1f75b031;rport;received=91.208.12.131

f: “weepee” sip:weepee@91.208.12.131;tag=as3ef22ef5

t: sip:s@81.82.255.37:5060;tag=as034b31a1

i: 06e6700b7dc1632c037f396d59853719@91.208.12.131:5060

CSeq: 102 OPTIONS

Server: taridium ipbx

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

k: replaces, timer

m: sip:81.82.255.37:5060

Accept: application/sdp

l: 0

<------------>
Scheduling destruction of SIP dialog ‘06e6700b7dc1632c037f396d59853719@91.208.12.131:5060’ in 32000 ms (Method: OPTIONS)
[Aug 27 09:48:13] e[1;33mNOTICEe[0m[25345]: e[1;37mchan_sip.ce[0m:e[1;37m13151e[0m e[1;37msip_reregistere[0m: – Re-registration for 329909014559@ssw9.brussels.weepee.org

e[Kipbx*CLI>
e[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.208.12.139:5060:
REGISTER sip:ssw9.brussels.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK24d786b3

Max-Forwards: 70

f: sip:329909014559@ssw9.brussels.weepee.org;tag=as177c4670

t: sip:329909014559@ssw9.brussels.weepee.org

i: 01c6fd81421b23f77047aacd4ddaea40@127.0.0.1

CSeq: 146 REGISTER

User-Agent: taridium ipbx

Authorization: Digest username=“329909014559”, realm=“weepee”, algorithm=MD5, uri=“sip:ssw9.brussels.weepee.org”, nonce=“2b6d2cf2”, response=“7801a5e313eeab4890895a35cae69148”

Expires: 30

m: sip:s@81.82.255.37:5060

l: 0

<— SIP read from UDP:91.208.12.131:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK126c8a7c;received=81.82.255.37;rport=5060

From: sip:329909014591@ssw1.brussels.weepee.org;tag=as22241ac1

To: sip:329909014591@ssw1.brussels.weepee.org;tag=as796f3c21

Call-ID: 3d971e8603d1ee695b04aa4933fa1e53@127.0.0.1

CSeq: 146 REGISTER

Server: weepee

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“weepee”, nonce=“7baf59c3”

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name ssw1.brussels.weepee.org

e[Kipbx*CLI>
e[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.208.12.131:5060:
REGISTER sip:ssw1.brussels.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK4944e601

Max-Forwards: 70

f: sip:329909014591@ssw1.brussels.weepee.org;tag=as3b668e0f

t: sip:329909014591@ssw1.brussels.weepee.org

i: 3d971e8603d1ee695b04aa4933fa1e53@127.0.0.1

CSeq: 147 REGISTER

User-Agent: taridium ipbx

Authorization: Digest username=“329909014591”, realm=“weepee”, algorithm=MD5, uri=“sip:ssw1.brussels.weepee.org”, nonce=“7baf59c3”, response=“916d2f657a284324f8d25f0f7b7e6050”

Expires: 30

m: sip:s@81.82.255.37:5060

l: 0

<— SIP read from UDP:91.208.12.139:5060 —>
OPTIONS sip:s@81.82.255.37:5060 SIP/2.0

Via: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK7d71ebbf;rport

Max-Forwards: 70

From: “weepee” sip:weepee@91.208.12.139;tag=as4defa14a

To: sip:s@81.82.255.37:5060

Contact: sip:weepee@91.208.12.139:5060

Call-ID: 6fbe52163ad5201e62cbbb580a384a6f@91.208.12.139:5060

CSeq: 102 OPTIONS

User-Agent: weepee

Date: Mon, 27 Aug 2012 08:01:06 GMT

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

Supported: replaces, timer

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.139:5060 —>
SIP/2.0 200 OK

v: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK7d71ebbf;rport;received=91.208.12.139

f: “weepee” sip:weepee@91.208.12.139;tag=as4defa14a

t: sip:s@81.82.255.37:5060;tag=as2ccf1ff6

i: 6fbe52163ad5201e62cbbb580a384a6f@91.208.12.139:5060

CSeq: 102 OPTIONS

Server: taridium ipbx

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

k: replaces, timer

m: sip:81.82.255.37:5060

Accept: application/sdp

l: 0

<------------>
Scheduling destruction of SIP dialog ‘6fbe52163ad5201e62cbbb580a384a6f@91.208.12.139:5060’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:91.208.12.139:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK24d786b3;received=81.82.255.37;rport=5060

From: sip:329909014559@ssw9.brussels.weepee.org;tag=as177c4670

To: sip:329909014559@ssw9.brussels.weepee.org;tag=as35a07195

Call-ID: 01c6fd81421b23f77047aacd4ddaea40@127.0.0.1

CSeq: 146 REGISTER

Server: weepee

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“weepee”, nonce=“68d4889a”

Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name ssw9.brussels.weepee.org

e[Kipbx*CLI>
e[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.208.12.139:5060:
REGISTER sip:ssw9.brussels.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK40b7e640

Max-Forwards: 70

f: sip:329909014559@ssw9.brussels.weepee.org;tag=as3614772d

t: sip:329909014559@ssw9.brussels.weepee.org

i: 01c6fd81421b23f77047aacd4ddaea40@127.0.0.1

CSeq: 147 REGISTER

User-Agent: taridium ipbx

Authorization: Digest username=“329909014559”, realm=“weepee”, algorithm=MD5, uri=“sip:ssw9.brussels.weepee.org”, nonce=“68d4889a”, response=“429b5fb1a0e0983077a9c5823928f8bd”

Expires: 30

m: sip:s@81.82.255.37:5060

l: 0

<— SIP read from UDP:91.208.12.131:5060 —>
OPTIONS sip:s@81.82.255.37:5060 SIP/2.0

Via: SIP/2.0/UDP 91.208.12.131:5060;branch=z9hG4bK26c81144;rport

Max-Forwards: 70

From: “weepee” sip:weepee@91.208.12.131;tag=as4cb5589b

To: sip:s@81.82.255.37:5060

Contact: sip:weepee@91.208.12.131:5060

Call-ID: 48b7d37b3e65cc0b6952c17520246c5c@91.208.12.131:5060

CSeq: 102 OPTIONS

User-Agent: weepee

Date: Mon, 27 Aug 2012 08:01:06 GMT

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

Supported: replaces, timer

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.131:5060 —>
SIP/2.0 200 OK

v: SIP/2.0/UDP 91.208.12.131:5060;branch=z9hG4bK26c81144;rport;received=91.208.12.131

f: “weepee” sip:weepee@91.208.12.131;tag=as4cb5589b

t: sip:s@81.82.255.37:5060;tag=as363ae3fd

i: 48b7d37b3e65cc0b6952c17520246c5c@91.208.12.131:5060

CSeq: 102 OPTIONS

Server: taridium ipbx

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

k: replaces, timer

m: sip:81.82.255.37:5060

Accept: application/sdp

l: 0

<------------>
Scheduling destruction of SIP dialog ‘48b7d37b3e65cc0b6952c17520246c5c@91.208.12.131:5060’ in 32000 ms (Method: OPTIONS)

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.131:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK4944e601;received=81.82.255.37;rport=5060

From: sip:329909014591@ssw1.brussels.weepee.org;tag=as3b668e0f

To: sip:329909014591@ssw1.brussels.weepee.org;tag=as796f3c21

Call-ID: 3d971e8603d1ee695b04aa4933fa1e53@127.0.0.1

CSeq: 147 REGISTER

Server: weepee

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

Supported: replaces, timer

Expires: 60

Contact: sip:s@81.82.255.37:5060;expires=60

Date: Mon, 27 Aug 2012 08:01:06 GMT

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘3d971e8603d1ee695b04aa4933fa1e53@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Aug 27 09:48:13] e[1;33mNOTICEe[0m[25345]: e[1;37mchan_sip.ce[0m:e[1;37m20905e[0m e[1;37mhandle_response_registere[0m: Outbound Registration: Expiry for ssw1.brussels.weepee.org is 60 sec (Scheduling reregistration in 45 s)

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.139:5060 —>
OPTIONS sip:s@81.82.255.37:5060 SIP/2.0

Via: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK22d35948;rport

Max-Forwards: 70

From: “weepee” sip:weepee@91.208.12.139;tag=as773d0f08

To: sip:s@81.82.255.37:5060

Contact: sip:weepee@91.208.12.139:5060

Call-ID: 0b347af93cf5fcf943b9d17662aa2b51@91.208.12.139:5060

CSeq: 102 OPTIONS

User-Agent: weepee

Date: Mon, 27 Aug 2012 08:01:07 GMT

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

Supported: replaces, timer

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in incoming (domain 81.82.255.37)

<— Transmitting (no NAT) to 91.208.12.139:5060 —>
SIP/2.0 200 OK

v: SIP/2.0/UDP 91.208.12.139:5060;branch=z9hG4bK22d35948;rport;received=91.208.12.139

f: “weepee” sip:weepee@91.208.12.139;tag=as773d0f08

t: sip:s@81.82.255.37:5060;tag=as6acd284f

i: 0b347af93cf5fcf943b9d17662aa2b51@91.208.12.139:5060

CSeq: 102 OPTIONS

Server: taridium ipbx

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

k: replaces, timer

m: sip:81.82.255.37:5060

Accept: application/sdp

l: 0

<------------>
Scheduling destruction of SIP dialog ‘0b347af93cf5fcf943b9d17662aa2b51@91.208.12.139:5060’ in 32000 ms (Method: OPTIONS)

e[Kipbx*CLI>
e[0KRetransmitting #1 (no NAT) to 91.208.12.139:5060:
REGISTER sip:ssw9.brussels.weepee.org SIP/2.0

v: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK40b7e640

Max-Forwards: 70

f: sip:329909014559@ssw9.brussels.weepee.org;tag=as3614772d

t: sip:329909014559@ssw9.brussels.weepee.org

i: 01c6fd81421b23f77047aacd4ddaea40@127.0.0.1

CSeq: 147 REGISTER

User-Agent: taridium ipbx

Authorization: Digest username=“329909014559”, realm=“weepee”, algorithm=MD5, uri=“sip:ssw9.brussels.weepee.org”, nonce=“68d4889a”, response=“429b5fb1a0e0983077a9c5823928f8bd”

Expires: 30

m: sip:s@81.82.255.37:5060

l: 0

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.139:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK40b7e640;received=81.82.255.37;rport=5060

From: sip:329909014559@ssw9.brussels.weepee.org;tag=as3614772d

To: sip:329909014559@ssw9.brussels.weepee.org;tag=as35a07195

Call-ID: 01c6fd81421b23f77047aacd4ddaea40@127.0.0.1

CSeq: 147 REGISTER

Server: weepee

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

Supported: replaces, timer

Expires: 60

Contact: sip:s@81.82.255.37:5060;expires=60

Date: Mon, 27 Aug 2012 08:01:07 GMT

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘01c6fd81421b23f77047aacd4ddaea40@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Aug 27 09:48:14] e[1;33mNOTICEe[0m[25345]: e[1;37mchan_sip.ce[0m:e[1;37m20905e[0m e[1;37mhandle_response_registere[0m: Outbound Registration: Expiry for ssw9.brussels.weepee.org is 60 sec (Scheduling reregistration in 45 s)

e[Kipbx*CLI>
e[0K
<— SIP read from UDP:91.208.12.139:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 81.82.255.37:5060;branch=z9hG4bK40b7e640;received=81.82.255.37;rport=5060

From: sip:329909014559@ssw9.brussels.weepee.org;tag=as3614772d

To: sip:329909014559@ssw9.brussels.weepee.org;tag=as35a07195

Call-ID: 01c6fd81421b23f77047aacd4ddaea40@127.0.0.1

CSeq: 147 REGISTER

Server: weepee

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“weepee”, nonce=“291729e0”, stale=true

Content-Length: 0

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

e[Kipbx*CLI> [/quote]

4

Did you buy a commercial product from Taridium? Are they providing you with commercial support?

5

It’s not a service unavailable problem, it is a “Forbidden” problem.

You mention two trunks. A lot of systems have trouble telling two SIP clients apart if they have the same IP address, on the other hand, you should not need two SIP trunks through the same provider.

6

We want to buy a commercial product from Taridium. But not before this is working. I contacted them already but they can’t/won’t really help us. They said to contact Weepee. So we did but we’re waiting for an appointment with the tech guy to do some live debugging. We’re waiting quite some time already…

The reason we have 2 SIP trunks through the same provider is because we’re with 2 different companies under one roof. The other company has their own phone numbers and their own account.
And to make it tricky, there’s one secretary who answers the phone for both companies. Also we may not be the same company but all the people need to be able make internal calls to everybody en transfer phone calls to everybody.

Would installing a second central be a solution? It would still have the same WAN IP. And it should be possible to make internal calls and transfers to everybody.

7

So i guess you have 2 accounts because you want a clean splitted bill for the outgoing calls.

This can be done by WeePee directly (more info in Dutch at nl.weepeetelecom.be/services/sip-trunk-account/)

Put the 2 companies in a separate context and then you can define for each company a specific way for the outgoing calls so that weepee can split the bills. Or maybe WeePee does the splitting based on the outgoing number.

8

[quote=“stijnpau”]We want to buy a commercial product from Taridium. But not before this is working. I contacted them already but they can’t/won’t really help us. They said to contact Weepee. So we did but we’re waiting for an appointment with the tech guy to do some live debugging. We’re waiting quite some time already…

The reason we have 2 SIP trunks through the same provider is because we’re with 2 different companies under one roof. The other company has their own phone numbers and their own account.
And to make it tricky, there’s one secretary who answers the phone for both companies. Also we may not be the same company but all the people need to be able make internal calls to everybody en transfer phone calls to everybody.

Would installing a second central be a solution? It would still have the same WAN IP. And it should be possible to make internal calls and transfers to everybody.[/quote]

Hello, please contact us at http://support.taridium.com if you need further help with your ipbx configuration.

thanks,
Mike

9

[quote=“tomdemoor”]So i guess you have 2 accounts because you want a clean splitted bill for the outgoing calls.

This can be done by WeePee directly (more info in Dutch at nl.weepeetelecom.be/services/sip-trunk-account/)

Put the 2 companies in a separate context and then you can define for each company a specific way for the outgoing calls so that weepee can split the bills. Or maybe WeePee does the splitting based on the outgoing number.[/quote]

That is indeed what we want. Any experience on that?
I haven’t been able to contact Weepee yet (been sick and all) but I hope to make progress on this issue this week.

@taridiumsupport
We already had an open ticket at your support service but I don’t believe I ever got an answer on my last mail. I just posted the log at your support pages.

10

The easiest thing is to let Weepee handle the splitted billing, it will be the most exact.
Else you can make 2 different contexts on which people dial out and have logging of the call (to which number and how long)