[Asterisk-Users] Getting started with Asterisk

[NOUR]

Hello ,

 

I'll just started with asterisk and I would liket to to dial between your
two phones with to cisco ATA 186 , but I have a problem 

 

The two cisco ATA and the server in the same networks and i have the ring in
the phone but i'am not able to place a call 

Between the twe phone .

 

In attachement the sip.conf and a log file

 

Any suggestement .

 

Regards

 

RAbii 

 

------------------------------sip.conf--------------------------------------
---------

 

[general]

 

port = 5060           ; Port to bind to (SIP is 5060)

bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)

allow=all             ; Allow all codecs

context = bogon-calls ; Send SIP callers that we don't know about here

 

[2000]

 

type=friend           ; This device takes and makes calls

username=2000         ; Username on device

secret=9overthruster7 ; Password for device

host=dynamic          ; This host is not on the same IP addr every time

context=from-sip      ; Inbound calls from this host go here

mailbox=100           ; Activate the message waiting light if this

dtmfmode=rfc2833                      ; voicemailbox has messages in it

 

[2001]                ; Duplicate of 2000, except with different auth data

 

type=friend

username=2001

secret=11bbanzai9

host=dynamic

context=from-sip

mailbox=101

dtmfmode=rfc2833

 

----------------------------------------------------------------------------
---------------------

~

 

 

 

 

-------------------------------Log
---------------------------------------------------

 

Asterisk*CLI> 

We're at 10.100.18.125 port 18294

Answering/Requesting with root capability 1

Answering with non-codec capability 0x1(G723)

12 headers, 10 lines

Reliably Transmitting:

INVITE sip:2001 at 10.100.18.124:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad

From: "NafthaChimie" <sip:2000 at 10.100.18.125>;tag=as49257c3d

To: <sip:2001 at 10.100.18.124:5060;user=phone;transport=udp>

Contact: <sip:2000 at 10.100.18.125>

Call-ID: 256db95b2c935d47464a5d0b7d9f82c4 at 10.100.18.125

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Wed, 01 Dec 2004 14:35:49 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Content-Type: application/sdp

Content-Length: 218

 

v=0

o=root 11153 11153 IN IP4 10.100.18.125

s=session

c=IN IP4 10.100.18.125

t=0 0

m=audio 18294 RTP/AVP 4 101

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

 (no NAT) to 10.100.18.124:5060

Asterisk*CLI> 

 

Sip read: 

SIP/2.0 488 Not Acceptable Here

Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad

From: "NafthaChimie" <sip:2000 at 10.100.18.125>;tag=as49257c3d

To: <sip:2001 at 10.100.18.124:5060;user=phone;transport=udp>;tag=556017164

Call-ID: 256db95b2c935d47464a5d0b7d9f82c4 at 10.100.18.125

CSeq: 102 INVITE

Server: Cisco ATA 186  v3.1.0 atasip (040211A)

Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER

Content-Length: 0

Warning: Media type not available

 

 

10 headers, 0 lines

Transmitting: 

ACK sip:2001 at 10.100.18.124:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad

From: "NafthaChimie" <sip:2000 at 10.100.18.125>;tag=as49257c3d

To: <sip:2001 at 10.100.18.124:5060;user=phone;transport=udp>;tag=556017164

Contact: <sip:2000 at 10.100.18.125>

Call-ID: 256db95b2c935d47464a5d0b7d9f82c4 at 10.100.18.125

CSeq: 102 ACK

User-Agent: Asterisk PBX

Content-Length: 0

 

 (no NAT) to 10.100.18.124:5060

Destroying call '256db95b2c935d47464a5d0b7d9f82c4 at 10.100.18.125'

 

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