No audio either way on remote extns

Asterisk Asterisk Support
21

A sip trace with… what?

Ethereal?
core set verbose?

Here you have a simple trace…
In this case, there is a X-Lite from one side. Because that is the change of port.

22

If you are not actually using NAT, take out the nat=route. I don’t know exactly what it will do, but it certainly won’t help.

23

just issue a SIP SET DEBUG on your asterisk CLI.

Post up the contents of the SIP transaction and then we can try and determine where the breakdown is.

24

Ok, thanks guys!

Watch please the last capture on the post number 33..
25

Give me a bit to sort through this. I have to distill it down into the various legs of the call to see if I can spot where the breakdown could possibly be.

26

After taking a look at things, I do see a problem which may be the source of this.

Here is the entire stream of SIP messaging between the two systems.

UA1 is the phone which issued the calling party, UA2 is the party you are trying to call.

UA1 --> AST : INVITE
AST --> UA1 : PROXY AUTH REQUIRED
UA1 --> AST : ACK
UA1 --> AST : INVITE (WITH PROXY AUTH)
AST --> UA1 : TRYING
AST --> UA2 : INVITE
UA2 --> AST : RINGING
AST --> UA1 : RINGING
UA2 --> AST : OK
AST --> UA2 : ACK
AST --> UA1 : OK
UA1 --> AST : ACK
UA2 --> AST : OK
AST --> UA2 : ACK
UA1 --> AST : BYE
AST --> UA1 : OK

UA2 re-sent the OK which makes me believe it never received the ACK from asterisk. I am not sure what this means, but there is definitely a breakdown in the messaging.

I assume 200.xx.xx.87 is the asterisk machine, correct? If so, it doesn’t appear that there is a RE-INVITE happening. That being said, I believe whatever the problem is has to do with UA2.

27

Check your user.conf file.

Look at the bottom of each user setup strings.

Check the following "disallow=all"
Check the following “allow =” “” for the right codecs.

You will prob find that both are set to “all”, correct it by changing the allow=codecs" or what ever codecs you are using.

users.conf takes priorty over sip.conf and extensions.conf.

Worth a try just to make sure it is correct.

28

I had a similar problem. The audio wouldn’t work and the SIP phones would only register for about 1 minute.

I found out that the problem was my Linksys router on the Asterisk Server side.

With the old DLINK (on the Server side) I had to forward my SIP ports (5060-5079) and RTP ports (10000-20000) to the Asterisk Box but I didn’t have to do any forwarding on the Client side (2-Wire Modem and also tried Siemens Gigaset).

When I changed the Router on the Asterisk side to a new Linksys, it wouldn’t work until I did the same port forwards on the Client Side.

You may want to try that.

David

29

[quote=“g2010”]
I assume 200.xx.xx.87 is the asterisk machine, correct?[/quote]
That’s correct!

[quote=“g2010”]
That being said, I believe whatever the problem is has to do with UA2.[/quote]
I don’t think so, because I tried this with many different UA2 and different internet providers

Sorry, but I don’t understand you very well, I’m using uLaw, aLaw and 729, so my sip.conf is:

sip.conf

disallow=all
allow=alaw
allow=ulaw
allow=g729

Should I set the same on users.conf?

Thanks newlinuxguy, but I already tried changing the Linksys routers. In fact, I tried without nothing between Sip phone and internet, just modem en bridge mode.

30

In web page of phone (page which is used for setups like server, user, callerID and password) there is also option STUN server.
And I put there
You can see more:
voip-info.org/wiki-STUN

31

Thanks bira_more, but right now I eliminated NAT from the extension phones, so I don’t need Stun ¿right?

32

I notice that UA1 is using private addresses in its INVITE. That’s naughty, although I don’t think it will confuse Asterisk, as Asterisk normally only checks the user part. Also it indicates that the description of the network is incomplete, in that the remote phones are connected via a proxy, not directly to the public network.

33

Yes, on the moment of capture, UA1 was behind a Linksys Router. Of course, the phone was using a invalid IP address.
Now, I changed that, and both phones are using public IP address and the result is the same.

UA1 ----> Asterisk <---- UA2

FYI: I got the same result with:

  1. Using Proxy (brekeke).
  2. Not-Using Proxy.
  3. Using Linksys Router with special firmware for VoIP.
  4. Not-Using Linksys on the middle.
  5. Hard-Phones.
  6. Softphones.
  7. Different HarPhones (Sysmaster (Tornado M5), Linksys (SPA941)

Right now, there is NO proxy, routers or anything in the middle. Just Phone1, Asterisk and Phone2…

I’m desperate!

34

Here is a more clean capture:

6001 = 200.xx.xx.199 —> X-Lite
6002 = 190.xx.xx.54 —> Linksys SPA941
Asterisk = 200.xx.xx.87

The call was: from 6002 to 6001

asterisco*CLI> sip set debug
SIP Debugging enabled
asterisco*CLI> 
<--- SIP read from 190.xx.xx.54:5060 --->
INVITE sip:6001@200.xx.xx.87 SIP/2.0
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-53c7e8e4
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "6002" <sip:6002@190.xx.xx.54:5060>
Expires: 240
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 394
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 526017 526017 IN IP4 190.xx.xx.54
s=-
c=IN IP4 190.xx.xx.54
t=0 0
m=audio 9002 RTP/AVP 18 0 2 4 8 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/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

<------------->
--- (13 headers 18 lines) ---
Sending to 190.xx.xx.54 : 5060 (no NAT)
Using INVITE request as basis request - 7cdf4c2c-b6d47b04@190.xx.xx.54

<--- Reliably Transmitting (no NAT) to 190.xx.xx.54:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-53c7e8e4;received=190.xx.xx.54
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>;tag=as09e7f7c7
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4428d80b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '7cdf4c2c-b6d47b04@190.xx.xx.54' in 32000 ms (Method: INVITE)
Found user '6002'

<--- SIP read from 190.xx.xx.54:5060 --->
ACK sip:6001@200.xx.xx.87 SIP/2.0
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-53c7e8e4
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>;tag=as09e7f7c7
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 101 ACK
Max-Forwards: 70
Contact: "6002" <sip:6002@190.xx.xx.54:5060>
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
asterisco*CLI> 
<--- SIP read from 190.xx.xx.54:5060 --->
INVITE sip:6001@200.xx.xx.87 SIP/2.0
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-347a4749
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="6002",realm="asterisk",nonce="4428d80b",uri="sip:6001@200.xx.xx.87",algorithm=MD5,response="3b
d52502d8371885ce96801a459f8c12"
Contact: "6002" <sip:6002@190.xx.xx.54:5060>
Expires: 240
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 394
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 526017 526017 IN IP4 190.xx.xx.54
s=-
c=IN IP4 190.xx.xx.54
t=0 0
m=audio 9002 RTP/AVP 18 0 2 4 8 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/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 190.xx.xx.54 : 5060 (no NAT)
Using INVITE request as basis request - 7cdf4c2c-b6d47b04@190.xx.xx.54
Found user '6002'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 190.xx.xx.54:9002
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
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 - 0x90e (gsm|ulaw|alaw|g726|g729), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x
90c (ulaw|alaw|g726|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 190.xx.xx.54:9002
Looking for 6001 in from-internal (domain 200.xx.xx.87)
list_route: hop: <sip:6002@190.xx.xx.54:5060>

<--- Transmitting (no NAT) to 190.xx.xx.54:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-347a4749;received=190.xx.xx.54
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001@200.xx.xx.87>
Content-Length: 0

<------------>
Audio is at 200.xx.xx.87 port 9034
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 200.xx.xx.199:58950:
INVITE sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5 SIP/2.0
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK562959e7;rport
From: "6002" <sip:6002@200.xx.xx.87>;tag=as405f8881
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>
Contact: <sip:6002@200.xx.xx.87>
Call-ID: 55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Jan 2009 12:43:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 362

v=0
o=root 7556 7556 IN IP4 200.xx.xx.87
s=session
c=IN IP4 200.xx.xx.87
t=0 0
m=audio 9034 RTP/AVP 0 18 111 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
asterisco*CLI> 
<--- Transmitting (no NAT) to 190.xx.xx.54:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-347a4749;received=190.xx.xx.54
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>;tag=as5554d350
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001@200.xx.xx.87>
Content-Length: 0

<------------>
asterisco*CLI> 
<--- SIP read from 200.xx.xx.199:58950 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK562959e7;rport=5060
Contact: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>;tag=e8354a54
From: "6002"<sip:6002@200.xx.xx.87>;tag=as405f8881
Call-ID: 55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87
CSeq: 102 INVITE
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
asterisco*CLI> 
<--- SIP read from 200.xx.xx.199:58950 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK562959e7;rport=5060
Contact: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>;tag=e8354a54
From: "6002"<sip:6002@200.xx.xx.87>;tag=as405f8881
Call-ID: 55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 191

v=0
o=- 6 2 IN IP4 200.xx.xx.199
s=CounterPath X-Lite 3.0
c=IN IP4 200.xx.xx.199
t=0 0
m=audio 36430 RTP/AVP 0 3 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 200.xx.xx.199:36430
Found audio description format telephone-event for ID 101
Capabilities: us - 0x90e (gsm|ulaw|alaw|g726|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|a
law)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 200.xx.xx.199:36430
list_route: hop: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>
set_destination: Parsing <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5> for address/port to send to
set_destination: set destination to 200.xx.xx.199, port 58950
Transmitting (no NAT) to 200.xx.xx.199:58950:
ACK sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5 SIP/2.0
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK6e89611d;rport
From: "6002" <sip:6002@200.xx.xx.87>;tag=as405f8881
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>;tag=e8354a54
Contact: <sip:6002@200.xx.xx.87>
Call-ID: 55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
Audio is at 200.xx.xx.87 port 9038
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
asterisco*CLI> 
<--- Reliably Transmitting (no NAT) to 190.xx.xx.54:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-347a4749;received=190.xx.xx.54
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>;tag=as5554d350
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001@200.xx.xx.87>
Content-Type: application/sdp
Content-Length: 335

v=0
o=root 7556 7556 IN IP4 200.xx.xx.87
s=session
c=IN IP4 200.xx.xx.87
t=0 0
m=audio 9038 RTP/AVP 0 18 2 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
asterisco*CLI> 
<--- SIP read from 190.xx.xx.54:5060 --->
ACK sip:6001@200.xx.xx.87 SIP/2.0
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-fc6df3de
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>;tag=as5554d350
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="6002",realm="asterisk",nonce="4428d80b",uri="sip:6001@200.xx.xx.87",algorithm=MD5,response="0d
3266460d6774698fa455b37e37eb5d"
Contact: "6002" <sip:6002@190.xx.xx.54:5060>
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
asterisco*CLI> 
<--- SIP read from 190.xx.xx.54:5060 --->
BYE sip:6001@200.xx.xx.87 SIP/2.0
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-a0daaa0
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>;tag=as5554d350
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest username="6002",realm="asterisk",nonce="4428d80b",uri="sip:6001@200.xx.xx.87",algorithm=MD5,response="d6
d1b592742d6f40f326c9a3443a01a9"
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 190.xx.xx.54 : 5060 (no NAT)

<--- Transmitting (no NAT) to 190.xx.xx.54:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.xx.xx.54:5060;branch=z9hG4bK-a0daaa0;received=190.xx.xx.54
From: "6002" <sip:6002@200.xx.xx.87>;tag=2b795a0c35dec6a4o0
To: "6001" <sip:6001@200.xx.xx.87>;tag=as5554d350
Call-ID: 7cdf4c2c-b6d47b04@190.xx.xx.54
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6001@200.xx.xx.87>
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87' in 7872 ms (Method: INVITE)
set_destination: Parsing <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5> for address/port to send to
set_destination: set destination to 200.xx.xx.199, port 58950
Reliably Transmitting (no NAT) to 200.xx.xx.199:58950:
BYE sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5 SIP/2.0
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK7dc2f648;rport
From: "6002" <sip:6002@200.xx.xx.87>;tag=as405f8881
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>;tag=e8354a54
Call-ID: 55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
asterisco*CLI> 
<--- SIP read from 200.xx.xx.199:58950 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK7dc2f648;rport=5060
Contact: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>;tag=e8354a54
From: "6002"<sip:6002@200.xx.xx.87>;tag=as405f8881
Call-ID: 55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87
CSeq: 103 BYE
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '55ed09da4a26aded0c103f7e5d47503e@200.xx.xx.87' Method: INVITE
Really destroying SIP dialog '7cdf4c2c-b6d47b04@190.xx.xx.54' Method: BYE
Reliably Transmitting (no NAT) to 200.xx.xx.199:58950:
OPTIONS sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5 SIP/2.0
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK66cd32d3;rport
From: "Unknown" <sip:Unknown@200.xx.xx.87>;tag=as135b3ed9
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>
Contact: <sip:Unknown@200.xx.xx.87>
Call-ID: 048d52a222d849db3dfb11b73fab3c78@200.xx.xx.87
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Jan 2009 12:44:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

---
asterisco*CLI> 
<--- SIP read from 200.xx.xx.199:58950 --->

<------------->
asterisco*CLI> 
<--- SIP read from 200.xx.xx.199:58950 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK66cd32d3;rport=5060
Contact: <sip:200.xx.xx.199:58950>
To: <sip:6001@200.xx.xx.199:58950;rinstance=e8119d13793b1af5>;tag=92486508
From: "Unknown"<sip:Unknown@200.xx.xx.87>;tag=as135b3ed9
Call-ID: 048d52a222d849db3dfb11b73fab3c78@200.xx.xx.87
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '048d52a222d849db3dfb11b73fab3c78@200.xx.xx.87' Method: OPTIONS
asterisco*CLI> sip set debug off
SIP Debugging Disabled
35

Watch please, the changes of RTP ports… Is that normal?

Should it use the same RTP ports all the time? I don’t know where else to search…

36

OK, next step here.

  • leave on SIP debugging
  • enable RTP debugging (rtp debug | rtp debug off)

Capture all the output to a file and post it on pastebin.com/ Let’s see who is currently sending audio through asterisk and who is not.

37

[quote=“g2010”]OK, next step here.

  • leave on SIP debugging
  • enable RTP debugging (rtp debug | rtp debug off)

Capture all the output to a file and post it on pastebin.com/ Let’s see who is currently sending audio through asterisk and who is not.[/quote]
THANKS g2010 !!!

I think that I saw what the problem is…

If I’m right, I will post the solution for this…

Thanks everybody!!!

38

please do elaborate!