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jafar mohammed

2005-Apr-03 17:37 UTC

[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@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@62.56.250.198>

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

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA@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@62.56.250.198>;tag=as08ade073

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

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:9001@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@62.56.250.198>;tag=as08ade073

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

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:9001@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@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@62.56.250.198>;tag=as08ade073

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

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA@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@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@62.56.250.198>;tag=as08ade073

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

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA@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@62.56.250.198>;tag=as08ade073

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

CSeq: 102 BYE

User-Agent: Asterisk PBX

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

Contact: <sip:9001@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@66.178.100.66'


		
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jafar mohammed

2005-Apr-03 17:41 UTC

head link

[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@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@62.56.250.198>

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

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA@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@62.56.250.198>;tag=as08ade073

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

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:9001@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@62.56.250.198>;tag=as08ade073

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

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:9001@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@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@62.56.250.198>;tag=as08ade073

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

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA@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@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@62.56.250.198>;tag=as08ade073

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

Call-ID:
ACD70110-A3D511D9-AA8D9D8C-BA1A49FA@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@62.56.250.198>;tag=as08ade073

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

CSeq: 102 BYE

User-Agent: Asterisk PBX

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

Contact: <sip:9001@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@66.178.100.66'

__________________________________________________
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jafar mohammed

2005-Apr-03 23:53 UTC

head link

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

AS5300 setup

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2005.04.04 09:37:31
=~=~=~=~=~=~=~=~=~=~=~sh runn
Building configuration...

Current configuration : 11599 bytes
!
! Last configuration change at 03:26:25 GMT Mon Apr 4
2005 by charles
! NVRAM config last updated at 03:06:50 GMT Mon Apr 4
2005 by charles
!
version 12.3
service timestamps debug datetime localtime
show-timezone
service timestamps log datetime localtime
show-timezone
service password-encryption
!
hostname 66.178.100.66
!
boot-start-marker
boot-end-marker
!
!
!
resource-pool enable
 --More--         !
resource-pool throttle 20 default
clock timezone GMT 3
clock calendar-valid
!
aaa new-model
!
!
aaa group server radius cdt-1
 server 159.148.8.108 auth-port 2362 acct-port 2363
!
aaa group server radius cdt-2
 server 62.85.77.82 auth-port 2362 acct-port 2363
!
aaa group server radius tsl
 server 62.56.250.200 auth-port 1812 acct-port 1813
!
aaa authentication login h323 group radius
aaa accounting send stop-record authentication failure

aaa accounting connection h323 stop-only broadcast
group cdt-1 group cdt-2 group tsl
aaa nas port voip
aaa session-id common
 --More--         ip subnet-zero
ip telnet source-interface FastEthernet0
ip name-server 66.178.100.68
!
!
!
trunk group  mgcp
!
isdn switch-type primary-net5
!
voice rtp send-recv
!
voice service pots 
!
voice service voip 
 cause-code legacy
 h323
  h225 timeout setup 8
  session transport udp
 sip
  min-se  600 
!
voice class codec 2
 --More--          codec preference 1 gsmfr
!
!
voice class permanent 1
 signal timing idle suppress-voice 5
 signal timing oos suppress-all 30
 signal timing oos timeout 120
!
!
voice class h323 1
 h225 timeout tcp establish 30
 h225 timeout connect 60
 h225 timeout setup 30
  call start fast
!
voice class h323 2
  call start slow
!
voice class h323 1001
  call start fast
!
voice class h323 10
!
 --More--         !
voice class busyout 1
!
!
voice class dualtone-detect-params 1
!
!
!
!
!
fax interface-type modem
!
!
controller E1 0
 clock source line primary
 ds0-group 0 timeslots 1-15,17-31 type r2-digital
!
!
!
translation-rule 22
 Rule 0 22254 254
!
!
!
interface Tunnel1
 ip address 192.168.44.1 255.255.255.0
 tunnel source Ethernet0
 tunnel destination 217.21.95.9
!
interface Tunnel17
 --More--          ip address 10.1.17.2 255.255.255.0
 shutdown
 tunnel source 212.165.147.254
 tunnel destination 66.92.133.199
 tunnel mode nos
!
interface Tunnel18
 no ip address
!
interface Ethernet0
 ip address 195.202.73.106 255.255.255.248
 no ip mroute-cache
!
interface Serial0
 no ip address
 no ip mroute-cache
 clockrate 2015232
 no fair-queue
!
interface Serial1
 no ip address
 no ip mroute-cache
 clockrate 2015232
 --More--          no fair-queue
!
interface Serial2
 no ip address
 no ip mroute-cache
 clockrate 2015232
 no fair-queue
!
interface Serial3
 no ip address
 no ip mroute-cache
 shutdown
 clockrate 2015232
 fair-queue 100 256 0
 ip rtp priority 10000 10000 75
!
interface Serial2:15
 no ip address
 isdn switch-type primary-net5
 no cdp enable
!
interface FastEthernet0
 ip address 172.16.202.90 255.255.255.0 secondary
 --More--          ip address 66.178.100.66
255.255.255.248
 ip access-group 1 in
 ip access-group 1 out
 no ip mroute-cache
 duplex auto
 speed auto
 h323-gateway voip interface
 h323-gateway voip id gk0 ipaddr 216.52.153.203 1719
 h323-gateway voip h323-id ngins
 ip rtp priority 16384 16383 400
!
ip classless
ip route 0.0.0.0 0.0.0.0 66.178.100.65
no ip http server
!
!
no logging trap
access-list 101 permit ip any any
!
route-map VOIP permit 20
 match ip address 101
!
route-map VOIP permit 100
 --More--         !
!
radius-server attribute 44 include-in-access-req
radius-server host 159.148.8.108 auth-port 2362
acct-port 2363 key 7 065E582A585C51411F0317
radius-server host 62.85.77.82 auth-port 2362
acct-port 2363 key 7 014B510F4F195E573B584B
radius-server host 62.56.250.200 auth-port 1812
acct-port 1813 key 7 121500031F0E050A
radius-server retransmit 10
radius-server timeout 120
radius-server vsa send accounting
radius-server vsa send authentication
call threshold global total-calls low 60 high 90
busyout
!
call application voice kenya flash:kenya.tcl
!
call application voice kenya1 flash:kenya.tcl
!

!
voice-port 0:0
 compand-type a-law
 connection plar 9001
!
!
mgcp call-agent 62.56.250.198 2427 service-type mgcp
version 1.0
mgcp dtmf-relay voip codec all mode out-of-band
mgcp restart-delay 2
mgcp codec g711ulaw packetization-period 10
mgcp package-capability dtmf-package
mgcp package-capability line-package
mgcp package-capability rtp-package
mgcp package-capability nas-package
mgcp package-capability script-package
mgcp sdp simple
 --More--         no mgcp validate domain-name
mgcp endpoint offset
mgcp bind control source-interface FastEthernet0
mgcp bind media source-interface FastEthernet0
mgcp behavior signals v0.1
!
mgcp profile default
!
dial-peer cor custom
!
!
!
!
dial-peer voice 271 pots
 permission orig
 huntstop
 application session
!
dial-peer voice 272 voip
 destination-pattern 9002
!
dial-peer voice 273 pots
 application session
!
dial-peer voice 270 voip
 max-conn 10
 incoming called-number 9002
 destination-pattern 9001
 progress_ind connect enable 8
 signaling forward unconditional
 --More--          voice-class sip rel1xx disable
 max-redirects 10
 session protocol sipv2
 session target ipv4:62.56.250.198
 session transport udp
 codec gsmfr
!
dial-peer voice 280 pots
 destination-pattern 9005
 port 0:0
!
gateway 
 resource threshold high 90 low 70 
!
sip-ua 
 srv version 1
 retry invite 1
 retry bye 1
 retry rel1xx 1
 timers trying 1000
 timers connect 1000
 timers disconnect 1000
 timers hold 15
 sip-server ipv4:62.56.250.198
!
banner login ^C^C
alias exec va sh call act v br
alias exec vh sh call his v br
alias exec ether sh interface fastethernet0
alias exec cont sh contr e1
alias exec voip sh voip rtp conn
alias exec port sh vo port summ
!
line con 0
line aux 0
line vty 0 4
 session-timeout 15
 exec-timeout 240 0
 password 7 045C1C535C711C1F
line vty 5
 password 7 083245431C4955
line vty 6 999
 password 7 10491E4C5647425A
!
ntp clock-period 17179619
ntp server 66.187.224.4
end


SIP config file

[general]
context=SIPCalls                        ; Default
context for incoming calls
;recordhistory=yes              ; Record SIP history
by default
                                ; (see sip history /
sip no history)
;realm=mydomain.tld             ; Realm for digest
authentication
                                ; defaults to
"asterisk"
                                ; Realms MUST be
globally unique according to RFC 3261
                                ; Set this to your
host name or domain name
port=5060                       ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind
to (0.0.0.0 binds to all)
srvlookup=no                    ; Enable DNS SRV
lookups on outbound calls
                                ; Note: Asterisk only
uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV
lookups disables the
                                ; ability to place SIP
calls based on domain
                                ; names to some other
SIP users on the Internet

pedantic=yes                    ; Enable slow,
pedantic checking for Pingtel
                                ; and multiline
formatted headers for strict
                                ; SIP compatibility
(defaults to "no")
tos=184                        ; Set IP QoS to either
a keyword or numeric val
tos=lowdelay                   ;
lowdelay,throughput,reliability,mincost,none
maxexpirey=3600         ; Max length of incoming
registration we allow
defaultexpirey=120              ; Default length of
incoming/outoing registration
notifymimetype=text/plain       ; Allow overriding of
mime type in MWI NOTIFY
;videosupport=yes               ; Turn on support for
SIP video

disallow=all                    ; First disallow all
codecs
allow=gsm
allow=ulaw
allow=alaw


[66.178.100.66]
type=user
context=AS5300
defaultip=66.178.100.66
nat=yes
dtmfmode=rfc2833
canreinvite=no



		
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asterisk users - Apr 2005 - AS5300+SIP+ASTERISK or AS5300+MGCP

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

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

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