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3.2.4. ipDialog SipTone 2 SIP hardphone

SipTone 2 is a harphone produced by ipDialog. You can find information also on the phones site http://www.ipdialog.com/products.htm. Here we will have a look at the phones menu and functions as well as using it to call throuhg Asterisk.

Note: Before going further make sure you have read how to configure Asterisk userasterisk

CONNECTING


To connect the phone plug one end of an Ethernet cable to the PC input and the other end to your PC LAN card. Plug the free end of your Internet cable to the phones Net input. The other end must be plugged where your Internet is coming from. Plug the adapter in its jack and connect it to the electricity network.


ADJUSTING


Now when you have connected the phone to the network you can add an user. Here is how the telephone looks like.



You can adjust the phone from the phone menu (to access it use the MENU/HANGUP Key) and also from the phones site in your local area network. We will use the site to adjust the phone because it is easier and more user friendly.

To see the IP address of your phone just click the INFO/MUTE Key of your phone. The second line you see on the phones display is the IP you look for. Now just type http://<Phone_IP_Address> and a dialog window prompting for username and password will appear.

Note: If your browser cannot open the site maybe the IP address is not in your local network. In this case just go to the phones menu choose 3)Settings/2)Phone Settings/2)Config Phone/3)Network/1)DHCP/1)Use DHCP. So now when you reboot the telephone it will have an IP from your local network. Type again the IP address in an internet browser and in the dialog window log with user:admin and password:admin. Below is the menu of the site.



1. General


This menu gives you general information for the phone like version of the phone, serial number and uptime.




2. Network Setup



In the dropdown field choose MANUAL and type the IP address you want for the telephone in the first text field (make sure the IP is not used by anybody). I choose 10.10.0.99 as you see. Note that after you change the IP address of the phone the site where you can access the setting will also be different. Gateway must be the IP address of your gateway (see you network settings). You also have to provide one or more IP for the DNS server that you use.


3. Phone Configuration



Here I register an user which in a while I will also add to Asterisk. You have to add some Full Name and User ID. It does not really matter what they are but it is prefferable to have some connection with the user using the phone. Dial plan field has to be filled with the number which will dial the phone. This number of course must be a valid extension in extensions.conf on Asterisk.


4. Servers



In REGISTAR choose MANUAL and in SIP URL add your Asterisk server in the following format: protocol (the phone supports just SIP), user, Asterisk IP Address sip:user@Asterisk_IP. In my case you see it is sip:ivan_new@10.3.3.25.

Remember: user ivan_new has to be added to sip.conf in Asterisk. Expire time is the time after which if nobody picks up the call will expire (here the default value is 3600sec which is one hour you). Server password must be the secret value of the user registered to Asterisk in sip.conf


5. Phonebook



In phonebook you can add contacts and keep them stored together.


6. Change Password



You can change the password which you use to log in. The dafault one is admin. Make sure you commit the change otherwise it will not have effect.


7. Advanced

Here you can make advanced changes to the phone. The ports for the SIP ans RTP are set by default as well as the Audio Codec (it is set to G.711 u-Law).





SETTING ASTERISK

To have the phone in work you have to add the user and extension you are going to use (or you already registered to the phone) in sip.conf and extensions.conf

1.sip.conf


Above is sip.conf and as you see I create user ivan_new of type friend (can call and can be called) with username ivan_new and the same password. I set the host IP to dynamic and add the user to the tutorial context.

2.extennions.conf

Here I just add the extension I declared in Phone Configuration to dial the phone. As you see it is number 6789 which dials ivan_new who is actualy registered on the phone.
exten => 6789,1,Dial(SIP/ivan_new)




For more information about how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk

 
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Jinho (skan1603 at empal dot com)
05 January 2006 12:33:09
We use two phone.
One is X-Lite. the Other is ipDialog.
When we make call from ipDialog to X-Lite. Call is Terminated, and ipDialog become Lockup state.

Please help me.

reference my sip debug log
*CLI>
*CLI>
<-- SIP read from 192.168.1.189:5060:
REGISTER sip:192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK72CF2788D8C34F68BFC2F25941114E73
From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Contact: "test" <sip:test@192.168.1.189:5060>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61434 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0


--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.189 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK72CF2788D8C34F68BFC2F25941114E73;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61434 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:test@192.168.1.180>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK72CF2788D8C34F68BFC2F25941114E73;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>;tag=as3406f1af
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61434 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:test@192.168.1.180>
WWW-Authenticate: Digest realm="asterisk", nonce="66ca7905"
Content-Length: 0


---
Scheduling destruction of call 'CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180' in 15000 ms

<-- SIP read from 192.168.1.189:5060:
REGISTER sip:192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK74F5F4C11D3C4710AF1E45949ECF6A18
From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Contact: "test" <sip:test@192.168.1.189:5060>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61435 REGISTER
Expires: 1800
Authorization: Digest username="test",realm="asterisk",nonce="66ca7905",response="f2aca0cc1962620bd3247387242ea1da",uri="sip:192.168.1.180"
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.189 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK74F5F4C11D3C4710AF1E45949ECF6A18;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61435 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:test@192.168.1.180>
Content-Length: 0


---
-- Registered SIP 'test' at 192.168.1.189 port 5060 expires 1800
-- Saved useragent "X-Lite release 1105x" for peer test
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK74F5F4C11D3C4710AF1E45949ECF6A18;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>;tag=as3406f1af
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61435 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 1800
Contact: <sip:test@192.168.1.189:5060>;expires=1800
Date: Thu, 05 Jan 2006 11:11:38 GMT
Content-Length: 0


---
Scheduling destruction of call 'CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180' in 15000 ms

<-- SIP read from 192.168.1.183:3072:
REGISTER sip:192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 00007878-f0f08888@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 26111 REGISTER
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00002a7f-f0f0da8f
Supported: timer
To: "ivan_new" <sip:ivan_new@192.168.1.180>
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Expires: 3600
Content-Length: 0


--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.183 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00002a7f-f0f0da8f
To: "ivan_new" <sip:ivan_new@192.168.1.180>
Call-ID: 00007878-f0f08888@192.168.1.183
CSeq: 26111 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:ivan_new@192.168.1.180>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00002a7f-f0f0da8f
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as1396cac1
Call-ID: 00007878-f0f08888@192.168.1.183
CSeq: 26111 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:ivan_new@192.168.1.180>
WWW-Authenticate: Digest realm="asterisk", nonce="0c9d913a"
Content-Length: 0


---
Scheduling destruction of call '00007878-f0f08888@192.168.1.183' in 15000 ms

<-- SIP read from 192.168.1.183:3073:
REGISTER sip:ivan_new@192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 00007878-f0f08888@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 26112 REGISTER
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=000022a0-f0f0d250
Supported: timer
To: "ivan_new" <sip:ivan_new@192.168.1.180>
Authorization: Digest username="ivan_new",realm="asterisk",uri="sip:ivan_new@192.168.1.180",nonce="0c9d913a",response="975e6b840b3be2c1605e82b92049872d"
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Expires: 3600
Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.183 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=000022a0-f0f0d250
To: "ivan_new" <sip:ivan_new@192.168.1.180>
Call-ID: 00007878-f0f08888@192.168.1.183
CSeq: 26112 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:ivan_new@192.168.1.180>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=000022a0-f0f0d250
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as1396cac1
Call-ID: 00007878-f0f08888@192.168.1.183
CSeq: 26112 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 3600
Contact: <sip:ivan_new@192.168.1.183>;expires=3600
Date: Thu, 05 Jan 2006 11:11:42 GMT
Content-Length: 0


---
Scheduling destruction of call '00007878-f0f08888@192.168.1.183' in 15000 ms

<-- SIP read from 192.168.1.189:5060:


--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 192.168.1.189:5060:


--- (0 headers 0 lines) Nat keepalive ---
Destroying call 'CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180'
Destroying call '00007878-f0f08888@192.168.1.183'

<-- SIP read from 192.168.1.189:5060:


--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 192.168.1.183:3074:
INVITE sip:4321@192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 000029b2-f0f0d942@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 28968 INVITE
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00004ee4-f0f0be14
Supported: timer
To: "4321" <sip:4321@192.168.1.180>
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY
Content-Type: application/sdp
Content-Length: 327

v=0
o=ivan_new 1009843207 1009843207 IN IP4 192.168.1.183
s=Sip Call
c=IN IP4 192.168.1.183
t=0 0
m=audio 5014 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (12 headers 16 lines)---
Using INVITE request as basis request - 000029b2-f0f0d942@192.168.1.183
Sending to 192.168.1.183 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00004ee4-f0f0be14
To: "4321" <sip:4321@192.168.1.180>;tag=as66239e6d
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4321@192.168.1.180>
Proxy-Authenticate: Digest realm="asterisk", nonce="0206f363"
Content-Length: 0


---
Scheduling destruction of call '000029b2-f0f0d942@192.168.1.183' in 15000 ms
Found user 'ivan_new'

<-- SIP read from 192.168.1.183:3074:
ACK sip:4321@192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 000029b2-f0f0d942@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 28968 ACK
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00004ee4-f0f0be14
To: "4321" <sip:4321@192.168.1.180>;tag=as66239e6d
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 192.168.1.183:3075:
INVITE sip:4321@192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 000029b2-f0f0d942@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 28969 INVITE
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Supported: timer
To: "4321" <sip:4321@192.168.1.180>
Proxy-Authorization: Digest username="ivan_new",realm="asterisk",uri="sip:4321@192.168.1.180",nonce="0206f363",response="31b7f0c87da6fcce97b81a78c2621514"
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY
Content-Type: application/sdp
Content-Length: 327

v=0
o=ivan_new 1009843207 1009843207 IN IP4 192.168.1.183
s=Sip Call
c=IN IP4 192.168.1.183
t=0 0
m=audio 5014 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (13 headers 16 lines)---
Using INVITE request as basis request - 000029b2-f0f0d942@192.168.1.183
Sending to 192.168.1.183 : 5060 (non-NAT)
Found user 'ivan_new'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.183:5014
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing),
combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4321 in tutorial (domain 192.168.1.180)
list_route: hop: <sip:ivan_new@192.168.1.183>
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28969 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4321@192.168.1.180>
Content-Length: 0


---
-- Executing Dial("SIP/ivan_new-caac", "SIP/test") in new stack
We're at 192.168.1.180 port 12938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.1.189:5060:
INVITE sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK57c33663;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 05 Jan 2006 11:12:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 4464 4464 IN IP4 192.168.1.180
s=session
c=IN IP4 192.168.1.180
t=0 0
m=audio 12938 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/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 - - - -

---
-- Called test

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK57c33663;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 102 INVITE
Server: X-Lite release 1105x
Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK57c33663;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 102 INVITE
Server: X-Lite release 1105x
Content-Length: 0


--- (9 headers 0 lines)---
-- SIP/test-73e2 is ringing
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28969 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4321@192.168.1.180>
Content-Length: 0


---

<-- SIP read from 192.168.1.189:5060:


--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK57c33663;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 102 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105x
Content-Length: 305

v=0
o=test 8711899 8714656 IN IP4 192.168.1.189
s=X-Lite
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.189:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:test@192.168.1.189:5060>
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Transmitting (no NAT) to 192.168.1.189:5060:
ACK sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK3edf6a0c;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/test-73e2 answered SIP/ivan_new-caac
We're at 192.168.1.180 port 17270
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28969 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4321@192.168.1.180>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 4464 4464 IN IP4 192.168.1.180
s=session
c=IN IP4 192.168.1.180
t=0 0
m=audio 17270 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Attempting native bridge of SIP/ivan_new-caac and SIP/test-73e2
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
We're at 192.168.1.180 port 12938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 14 lines
Reliably Transmitting (no NAT) to 192.168.1.189:5060:
INVITE sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK64a72bf2;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 4464 4465 IN IP4 192.168.1.183
s=session
c=IN IP4 192.168.1.183
t=0 0
m=audio 5014 RTP/AVP 0 4 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/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 - - - -

---

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK64a72bf2;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 103 INVITE
Server: X-Lite release 1105x
Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK64a72bf2;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 103 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105x
Content-Length: 305

v=0
o=test 8711899 8714656 IN IP4 192.168.1.189
s=X-Lite
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.189:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Transmitting (no NAT) to 192.168.1.189:5060:
ACK sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK77326121;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 192.168.1.183:3075:
ACK sip:4321@192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 000029b2-f0f0d942@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 28969 ACK
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Content-Length: 0


--- (9 headers 0 lines)---
set_destination: Parsing <sip:ivan_new@192.168.1.183> for address/port to send to
set_destination: set destination to 192.168.1.183, port 5060
We're at 192.168.1.180 port 17270
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 14 lines
Reliably Transmitting (no NAT) to 192.168.1.183:5060:
INVITE sip:ivan_new@192.168.1.183 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK6cf0e90e;rport
From: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Contact: <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 4464 4465 IN IP4 192.168.1.189
s=session
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 192.168.1.183:3075:
BYE sip:4321@192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28970 BYE
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Supported: timer
To: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
Proxy-Authorization: Digest username="ivan_new",realm="asterisk",uri="sip:4321@192.168.1.180",nonce="0206f363",response="930bacba2a9bec82d04a886decbaf0bd"
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 192.168.1.183 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28970 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4321@192.168.1.180>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
We're at 192.168.1.180 port 12938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.1.189:5060:
INVITE sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK5c251319;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 4464 4466 IN IP4 192.168.1.180
s=session
c=IN IP4 192.168.1.180
t=0 0
m=audio 12938 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/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 - - - -

---
== Spawn extension (tutorial, 4321, 1) exited non-zero on 'SIP/ivan_new-caac'

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK5c251319;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 104 INVITE
Server: X-Lite release 1105x
Content-Length: 0


--- (9 headers 0 lines)---

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK5c251319;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 104 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105x
Content-Length: 305

v=0
o=test 8711899 8714656 IN IP4 192.168.1.189
s=X-Lite
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.189:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Transmitting (no NAT) to 192.168.1.189:5060:
ACK sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK459d3856;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Reliably Transmitting (no NAT) to 192.168.1.189:5060:
BYE sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK0fd45348;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 192.168.1.189:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK0fd45348;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 105 BYE
Server: X-Lite release 1105x
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180'
Retransmitting #1 (no NAT) to 192.168.1.183:5060:
INVITE sip:ivan_new@192.168.1.183 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK6cf0e90e;rport
From: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Contact: <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 4464 4465 IN IP4 192.168.1.189
s=session
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #2 (no NAT) to 192.168.1.183:5060:
INVITE sip:ivan_new@192.168.1.183 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK6cf0e90e;rport
From: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Contact: <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 4464 4465 IN IP4 192.168.1.189
s=session
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #3 (no NAT) to 192.168.1.183:5060:
INVITE sip:ivan_new@192.168.1.183 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK6cf0e90e;rport
From: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Contact: <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 4464 4465 IN IP4 192.168.1.189
s=session
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #4 (no NAT) to 192.168.1.183:5060:
INVITE sip:ivan_new@192.168.1.183 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK6cf0e90e;rport
From: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Contact: <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 4464 4465 IN IP4 192.168.1.189
s=session
c=IN IP4 192.168.1.189
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---






 
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