Asterisk 1.4 with eyeBeam softphones over VPN problem

cjrb2009 · March 5, 2013, 3:44am

Hi everyone,
I have searched the forum but couldn’t find the answer to my problem.
So the set up is following:
Asterisk 1.4 with Asterisk GUI is running in one location. Network is 192.168.1.0
2 softphones (Eyebeam) installed on the computers that are connected to the remote location where asterisk is hosted via VPN. Network where softphones are installed is 192.168.0.0 Softphones have extentions 100 and 102. Everything works but voicemail. When user at 100 dial voicemail for some reason asterisk is trying to log it in to the box for 102 thus creating problem with passwords. User at 102 can login into voice mail without a problem to the right box. If user 102 is signed out then user 100 can access it’s mail box with no problem.
In the system users 100 and 102 registered at 192.168.1.1 but on different ports.

So what I don’t get is that users 100 and 102 have no issues with incoming and outgoing call, calls between each other and with users in remote network (192.168.1.0), the only problem is that for some reason when one user tries to access voice mail astersik thinks that it is another user?! How come? Can it be related that both users for some reason registered under the same IP? But then wouldn’t calling be affected also?

Any suggestions are appreciated. It just drives me crazy.
BTW I also added second localnet=192.168.0.0/255.255.255.0 I have read on the forum.

Thanks in advance. I hope for the help.

dejanst · March 5, 2013, 10:03am

Did you also set externip= parameter?

Please do the following debugs via Asterisk CLI and copy/paste the output here:

sip show peers (when all the clients are registered to Asterisk)

sip set debug on (and make a call from remote phone to voicemail number)

make sure that Verbosity in Asterisk CLI is set to 3 or higher.

cjrb2009 · March 6, 2013, 1:18am

[quote=“dejanst”]Did you also set externip= parameter?

Please do the following debugs via Asterisk CLI and copy/paste the output here:

sip show peers (when all the clients are registered to Asterisk)

sip set debug on (and make a call from remote phone to voicemail number)

make sure that Verbosity in Asterisk CLI is set to 3 or higher.[/quote]

dejanst thank you for your reply. So the problem is with user 101 and 104. When user 104 is logged in user 101 can’t access its mail box, instead it tries to access 104 mail box.

Yes, externip is set to our external IP address. Asterisk is hosted on 192.168.1.5
So, sip show peers is:

Name/username Host Dyn Nat ACL Port Status
105/105 (Unspecified) D N 0 UNKNOWN
104/104 192.168.1.1 D N 40032 OK (139 ms)
152/152 (Unspecified) D N 0 UNKNOWN
155/155 (Unspecified) D N 0 UNKNOWN
151/151 (Unspecified) D N 0 UNKNOWN
150/150 192.168.1.45 D N 5060 OK (10 ms)
102/102 (Unspecified) D N 0 UNKNOWN
101/101 192.168.1.1 D N 10101 OK (136 ms)
100/100 192.168.1.46 D N 5060 OK (108 ms)

I have enabled sip debug on and got a lot of data. Unfortunately I can’t read it, so I’m going to post it here.
Debug:

<------------->
— (13 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.1 : 10101 (NAT)
Found peer '101’
Looking for 101 in device-hints (domain 192.168.1.5)

<— Transmitting (NAT) to 192.168.1.1:10101 —>
SIP/2.0 404 Not Found
v: SIP/2.0/UDP 192.168.1.1:10101;branch=z9hG4bK-d87543-281aa23688246131-1–d87543-;received=192.168.1.1;rport=10101
f: "101"sip:101@192.168.1.5:5060;tag=c01a8c03
t: "101"sip:101@192.168.1.5:5060;tag=as17307e49
i: f37d7c224a1b7b43MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0

<------------>
Really destroying SIP dialog ‘f37d7c224a1b7b43MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.’ Method: SUBSCRIBE
Reliably Transmitting (NAT) to 192.168.1.1:10101:
OPTIONS sip:101@192.168.1.1:10101;rinstance=64aee8b86ead325b SIP/2.0
v: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK3c36838e;rport
f: “asterisk” sip:asterisk@192.168.1.5;tag=as7c085259
t: sip:101@192.168.1.1:10101;rinstance=64aee8b86ead325b
m: sip:asterisk@192.168.1.5
i: 19d3ea40197b35fc0d3a1e0e5a1395ab@192.168.1.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 06 Mar 2013 00:53:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0

<— SIP read from 192.168.1.1:10101 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK3c36838e;rport=5060
Contact: sip:192.168.1.1:10101
To: sip:101@192.168.1.1:10101;rinstance=64aee8b86ead325b;tag=f724e50f
From: "asterisk"sip:asterisk@192.168.1.5:5060;tag=as7c085259
Call-ID: 19d3ea40197b35fc0d3a1e0e5a1395ab@192.168.1.5
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘19d3ea40197b35fc0d3a1e0e5a1395ab@192.168.1.5’ Method: OPTIONS

<— SIP read from 192.168.1.1:10101 —>
INVITE sip:700@192.168.1.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:10101;branch=z9hG4bK-d87543-65601c712f1d0f30-1–d87543-;rport
Max-Forwards: 70
Contact: sip:101@192.168.1.1:10101
To: "700"sip:700@192.168.1.5:5060
From: "101"sip:101@192.168.1.5:5060;tag=44400d66
Call-ID: ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 494

v=0
o=- 9 2 IN IP4 192.168.1.1
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.1
t=0 0
m=audio 10164 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101
a=alt:1 2 : GCAItlmk z5JCqY4h 192.168.0.220 10164
a=alt:2 1 : oXcKC6Li 4pwypvbN 192.168.1.51 10164
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:101 telephone-event/8000
=sendrecvory*CLI>
a=x-rtp-session-id:00D6BA599CFE4CF39F4E004E2985567A

<------------->
— (12 headers 17 lines) —
Sending to 192.168.1.1 : 10101 (NAT)
Using INVITE request as basis request - ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
Found peer '104’
Found RTP audio format 107
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.1:10164
Found unknown media description format BV32 for ID 107
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0x20f (g723|gsm|ulaw|alaw|speex), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x20e (gsm|ulaw|alaw|speex)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.1:10164
Looking for 700 in DLPN_JOHN (domain 192.168.1.5)
list_route: hop: sip:101@192.168.1.1:10101

<— Transmitting (NAT) to 192.168.1.1:10101 —>
SIP/2.0 100 Trying
v: SIP/2.0/UDP 192.168.1.1:10101;branch=z9hG4bK-d87543-65601c712f1d0f30-1–d87543-;received=192.168.1.1;rport=10101
f: "101"sip:101@192.168.1.5:5060;tag=44400d66
t: "700"sip:700@192.168.1.5:5060
i: ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
m: sip:700@192.168.1.5
l: 0

<------------>
– Executing [700@DLPN_JOHN:1] VoiceMailMain(“SIP/104-1078aa98”, “104@default”) in new stack
Audio is at 192.168.1.5 port 15970
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x200 (speex) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.1:10101 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.1.1:10101;branch=z9hG4bK-d87543-65601c712f1d0f30-1–d87543-;received=192.168.1.1;rport=10101
f: "101"sip:101@192.168.1.5:5060;tag=44400d66
t: "700"sip:700@192.168.1.5:5060;tag=as27a982b0
i: ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
m: sip:700@192.168.1.5
c: application/sdp
l: 312

v=0
o=root 5109 5109 IN IP4 192.168.1.5
s=session
c=IN IP4 192.168.1.5
t=0 0
m=audio 15970 RTP/AVP 0 3 8 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– <SIP/104-1078aa98> Playing ‘vm-password’ (language ‘en’)
john*CLI>
<— SIP read from 192.168.1.1:10101 —>
ACK sip:700@192.168.1.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:10101;branch=z9hG4bK-d87543-810cec752616847a-1–d87543-;rport
Max-Forwards: 70
Contact: sip:101@192.168.1.1:10101
To: "700"sip:700@192.168.1.5:5060;tag=as27a982b0
From: "101"sip:101@192.168.1.5:5060;tag=44400d66
Call-ID: ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
CSeq: 1 ACK
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Incorrect password ‘0000’ for user ‘104’ (context = default)
– <SIP/104-1078aa98> Playing ‘vm-incorrect’ (language ‘en’)
– <SIP/104-1078aa98> Playing ‘vm-password’ (language ‘en’)
john*CLI>
<— SIP read from 192.168.1.1:10101 —>
BYE sip:700@192.168.1.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:10101;branch=z9hG4bK-d87543-f903e709a078e777-1–d87543-;rport
Max-Forwards: 70
Contact: sip:101@192.168.1.1:10101
To: "700"sip:700@192.168.1.5:5060;tag=as27a982b0
From: "101"sip:101@192.168.1.5:5060;tag=44400d66
Call-ID: ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
CSeq: 2 BYE
User-Agent: eyeBeam release 1003s stamp 31159
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.1 : 10101 (NAT)

<— Transmitting (NAT) to 192.168.1.1:10101 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.1.1:10101;branch=z9hG4bK-d87543-f903e709a078e777-1–d87543-;received=192.168.1.1;rport=10101
f: "101"sip:101@192.168.1.5:5060;tag=44400d66
t: "700"sip:700@192.168.1.5:5060;tag=as27a982b0
i: ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
m: sip:700@192.168.1.5
l: 0

<------------>
[Mar 5 16:53:50] WARNING[10036]: app_voicemail.c:6328 vm_authenticate: Unable to read password
Really destroying SIP dialog ‘ba633a460d588075MTFmZjlmNjhhZmI2YjhiNWI4ZWRhMjE4ZDdjODNhNjU.’ Method: BYE

So I have no idea why is it doing this.
Any help is very much appreciated.
Thank you in advance.