[Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP

Sun Apr 3 20:08:54 MST 2005

If u want some help put your 53xx and sip config files.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jafar mohammed
Sent: Sunday, April 03, 2005 9:41 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP

hi's

i have been trying to configure my AS5300 to work with my asterisk box. i
have tried SIP, calls come, answered and AS5300 sends BYE message after not
more than 5 secs. I have also tried MGCP, but i believe i am not configuring
that right. here is the output of the sip debug. please help me out or lead
me to the direction of sorting this problem out.

thank you

INVITE sip:9001 at 62.56.250.198:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: <sip:66.178.100.66>;tag=8CB7504-1904

To: <sip:9001 at 62.56.250.198>

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA at 66.178.100.66

Supported: timer

Min-SE:  600

Cisco-Guid:
2899651584-2748649945-2861211020-3122285050

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Max-Forwards: 6

Remote-Party-ID:
<sip:66.178.100.66>;party=calling;screen=no;privacy=off

Timestamp: 1112573810

Contact: <sip:66.178.100.66:5060>

Expires: 180

Allow-Events: telephone-event

MIME-Version: 1.0

Content-Type: multipart/mixed;boundary=uniqueBoundary

Content-Length: 431

--uniqueBoundary

Content-Type: application/sdp

v=0

o=CiscoSystemsSIP-GW-UserAgent 5042 571 IN IP4
66.178.100.66

s=SIP Call

c=IN IP4 66.178.100.66

t=0 0

m=audio 18992 RTP/AVP 3 19

c=IN IP4 66.178.100.66

a=rtpmap:3 GSM/8000

a=rtpmap:19 CN/8000

a=ptime:10

--uniqueBoundary

Content-Type: application/gtd

Content-Disposition: signal;handling=optional

IAM,

GCI,acd52c00a3d511d9aa8a9d8cba1a49fa

--uniqueBoundary--

-*-

   - 21 headers, 21 lines

* Using latest SIP request as basis request

* Sending to 66.178.100.66 : 5060 (NAT)
Apr  4 00:16:40 NOTICE[25296]: chan_sip2.c:5872
check_user_full: User name from URI: 66.178.100.66, Digest auth user: (null)

  == Authentication turned off, no secret for user
66.178.100.66

* No RDNIS header in SIP packet

    -- - SIPFromURI:
<sip:66.178.100.66>;tag=8CB7504-1904

-->  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 66.178.100.66:5060

From: <sip:66.178.100.66>;tag=8CB7504-1904

To: <sip:9001 at 62.56.250.198>;tag=as08ade073

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA at 66.178.100.66

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:9001 at 62.56.250.198>

Content-Length: 0

-*-

    -- Executing Answer("SIP/66.178.100.66-bf34", "") in new stack

* SDP preparation: We're at 62.56.250.198 port 17962

* Answering with preferred capability 0x2 (gsm)

* Answering with preferred capability 0x4 (ulaw)

* Answering with preferred capability 0x8 (alaw)

Answering with non-codec capability 0x1(g723)

--> Reliably  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.178.100.66:5060

From: <sip:66.178.100.66>;tag=8CB7504-1904

To: <sip:9001 at 62.56.250.198>;tag=as08ade073

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA at 66.178.100.66

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:9001 at 62.56.250.198>

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 25296 25296 IN IP4 62.56.250.198

s=session

c=IN IP4 62.56.250.198

t=0 0

m=audio 17962 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 - - - -

-*-

    -- Executing Wait("SIP/66.178.100.66-bf34", "2") in new stack

--- Sip read from 66.178.100.66:50341
ACK sip:9001 at 62.56.250.198:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: <sip:66.178.100.66>;tag=8CB7504-1904

To: <sip:9001 at 62.56.250.198>;tag=as08ade073

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA at 66.178.100.66

Max-Forwards: 6

Content-Length: 0

CSeq: 101 ACK

-*-

   - 9 headers, 0 lines

--- Sip read from 66.178.100.66:53065
BYE sip:9001 at 62.56.250.198:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: <sip:66.178.100.66>;tag=8CB7504-1904

To: <sip:9001 at 62.56.250.198>;tag=as08ade073

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA at 66.178.100.66

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 6

Timestamp: 1112573810

CSeq: 102 BYE

Content-Length: 0

-*-

   - 11 headers, 0 lines

* Sending to 66.178.100.66 : 5060 (non-NAT)

-->  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.178.100.66:5060

From: <sip:66.178.100.66>;tag=8CB7504-1904

To: <sip:9001 at 62.56.250.198>;tag=as08ade073

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA at 66.178.100.66

CSeq: 102 BYE

User-Agent: Asterisk PBX

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

Contact: <sip:9001 at 62.56.250.198>

Content-Length: 0

-*-

  == Spawn extension (AS5300, 9001, 2) exited non-zero on
'SIP/66.178.100.66-bf34'

    -- Executing Hangup("SIP/66.178.100.66-bf34", "") in new stack

  == Spawn extension (AS5300, h, 1) exited non-zero on
'SIP/66.178.100.66-bf34'

Destroying SIP dialogue 'ACD70110-A3D511D9-AA8D9D8C-BA1A49FA at 66.178.100.66'

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