patton 4634 only two simultaneous call allowed

[luppazzo]

I have a patton 4634 3bri and 2 ISDN lines in point-to-point configuration

when receiving calls I get up to 4 simultaneous calls
when dialing I am only allowed to do up to 2 simultaneous call

any ide where to cahnge gateway configuration?

this is the patton status:
Domain: 192.168.0.2
Local Address: 192.168.0.7:5060
Remote Address: (none):0
Default Server: 192.168.0.2:0;lr
Early Proceeding: yes
Hold using zero IP: yes
Early Connect: no
Early Disconnect: no
Call-Transfer: accept & emit
Call-Reroute: disabled
AOC-D Information: accept
Pull-In Externally Looped Calls: yes
Address-Complete Indication: Accept: clear
PhoneContext
Use Calling Remote-Party-ID: no
Use Called Remote-Party-ID: yes
Use Privacy: no

Outgoing Call Parameter Source
------------------------------

From Header
User: from call URI or E.164 ()
Host-Port: from call URI or E.164 0)

To Header
User: from call URI or E.164 ()
Host-Port: from interface 0)

Request URI
User: fix (10001)
Host-Port: from To header 0)
Target Parameter: none (Target: , Cause: )

Calling Remote-Party-ID Header
Header User: from call URI or E.164 ()
Header Host-Port: from call URI or E.164 0)

Called Remote-Party-ID Header
Header User: from call URI or E.164 ()
Header Host-Port: from interface 0)

Identity Header
Header User: from call URI or E.164 ()
Header Host-Port: from interface 0)

Diversion Header
Header Host-Port: none 0)

Incoming Call Parameter Source
------------------------------

Calling Party
E.164: from Identity header ()
URI: from Identity header ((invalid))
Name: from Identity header ()
Redirecting E.164: none (Target: , Cause: Unknown)

Called Party
E.164: from Request-URL ()
URI: from Remote-Party-ID header ((invalid))
Name: from Remote-Party-ID header ()
Redirecting E.164: none (Target: , Cause: Unknown)

Play Tones: yes

VoIP Profile: default
---------------------

Used: by 2 module(s)

Voice Codecs
G.711 A-law: Rx=20ms Tx=20ms
G.711 u-law: Rx=20ms Tx=20ms

Fax Codecs
T.38 UDP: Rx=1ms Tx=1ms Relay

Modem Codecs
Codec Negotiation: enabled

Dejitter
Mode: Adaptive
Max. Delay: 60ms
Max. Packet Loss: 4/1000
Shrink Speed: 1
Grow Step: 1
Grow Attenuation: 1
High Pass Filter: enabled
Post Filter: enabled

Fax
Detection: CED Tone
T.38 High Speed Redundant Packets: 0
T.38 Low Speed Redundant Packets: 0
Max. Bit Rate: 14400bps
Volume: -9.500dB
Error Correction: enabled
HDLC: enabled
Dejitter Max Delay: 200ms

Modem
Max. Bit Rate: 14400
Volume: -9.500dB
HDLC: enabled
Dejitter Max Delay: 200ms

DTMF
Relay: enabled
Mute Encoder: enabled

RTP
Payload Type NTE: 101
Payload Type NSE: 100
Payload Type Clearmode: 97
Payload Type Redundancy: 255
Payload Type G.726-32: 2
Payload Type G.726-32-AAL2: 2

BCD
Avtivation: on-rx-activation
Timeout: 1000
Traffic Class: local-voice

SIP Profile: default
--------------------

Used: by 2 module(s)
Autonomous Transitioning: disabled

Tone-Set Profile: (none)
Routing Destination: router (IF_SIP_0-precall-service)
State: Up
Allow New Calls: yes
Allow Incoming Calls: yes
Wait For Name: no
Wait For Number: no
Proceeding Timeout: 4000
Early Alerting Timeout: 0

Endpoints
---------

Endpoint: IF_SIP_0-00b69fd8
Domain: 192.168.0.2
Local Host: 192.168.0.7
Local Port: 5060
Remote Host: (none)
Remote Port: (default)
Default Server: 192.168.0.2/0 (loose-router)
Session-Timer Interval: 0s
Session-Timer Version: 8
Call-Transfer Version: 5
Early Proceeding: yes
Hold using zero IP: yes
Early Connect: no
Early Disconnect: no
Call-Transfer: accept & emit
Call-Reroute: disabled
AOC-D Information: accept
Pull-In Externally Looped Calls: yes
Address-Complete Indication: Accept: clear
Phone Context
Use Calling Remote-Party-ID: no
Use Called Remote-Party-ID: yes
Play Tones: yes
Tone-Set Profile: (none)
Call Routing Destination: router (IF_SIP_0-precall-service)
Active State: Disconnected
Passive State: Idle

Endpoint: IF_SIP_0-009297b8
Domain: 192.168.0.2
Local Host: 192.168.0.7
Local Port: 5060
Remote Host: (none)
Remote Port: (default)
Default Server: 192.168.0.2/0 (loose-router)
Session-Timer Interval: 0s
Session-Timer Version: 8
Call-Transfer Version: 5
Early Proceeding: yes
Hold using zero IP: yes
Early Connect: no
Early Disconnect: no
Call-Transfer: accept & emit
Call-Reroute: disabled
AOC-D Information: accept
Pull-In Externally Looped Calls: yes
Address-Complete Indication: Accept: clear
Phone Context
Use Calling Remote-Party-ID: no
Use Called Remote-Party-ID: yes
Play Tones: yes
Tone-Set Profile: (none)
Call Routing Destination: router (IF_SIP_0-precall-service)
Active State: Connected
Passive State: Idle

 

[mfm]

Hi,

On the Management console, select the Patton 4634 Gateway, and in the outbound parameters tab kindly ensure they are set to "DevHostPort", if you are unusure kindly tell me what they are.

RequestLineURI : Host Part
To : Host Part

Thanks,

 

[luppazzo]

thank you for your prompt reply

I have on the config
Sip fileds:
Request Line URI : Host Part "DevHostPort" source address/port
To : Host Part "DevHostPort" source address/port

Here's the config file for further check; can you see any other possible issue?
Another technician suggested the problem was in the pp pmp configuration of ISDN line but I doublechecked with telco and they are all correct.


#########################################################
#
# For SN-4634/3BIS/UI 3-port BRI
#
# Author: Kevin Attard Compagno
# Date: 2008-02-20
# Contact: [email protected]
#
#########################################################

webserver port 80 language en

system

ic voice 0
low-bitrate-codec g729

system
clock-source 0 0

profile ppp default

profile call-progress-tone defaultDialtone
flush-play-list
play 1 200 425 0
pause 2 200
play 1 600 425 0
pause 1 1000

profile call-progress-tone defaultAlertingtone
flush-play-list
play 1 1000 425 -7
pause 2 4000

profile call-progress-tone defaultBusytone
flush-play-list
play 1 200 425 -7
pause 2 200

profile call-progress-tone defaultReleasetone
flush-play-list
play 1 250 425 -7
pause 2 250

profile call-progress-tone defaultCongestiontone
flush-play-list
play 1 250 425 -7
pause 2 250

profile tone-set default

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
fax transmission 1 relay t38-udp

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_WAN
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface IF_IP_LAN
ipaddress 192.168.0.7 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context cs switch

mapping-table calling-pi to calling-e164 MP_REM_CLIR
map restricted to ""

routing-table called-e164 RT_ISDN_TO_SIP_0
route T2 dest-interface IF_SIP_0 MP_REM_CLIR

routing-table called-e164 RT_ISDN_TO_SIP_1
route T2 dest-interface IF_SIP_1 MP_REM_CLIR

interface isdn IF_ISDN_0
route call dest-table RT_ISDN_TO_SIP_0

interface isdn IF_ISDN_1
route call dest-table RT_ISDN_TO_SIP_1

interface sip IF_SIP_0
bind gateway GW_SIP_0
%%earlymedia%%
service default
route call dest-interface IF_ISDN_0
remote-party-id called-party
address-translation outgoing-call request-uri user-part fix 10001 host-part to-header target-param none
address-translation incoming-call called-e164 request-uri

interface sip IF_SIP_1
bind gateway GW_SIP_1
%%earlymedia%%
service default
route call dest-interface IF_ISDN_1
remote-party-id called-party
address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none
address-translation incoming-call called-e164 request-uri

context cs switch
no shutdown

gateway sip GW_SIP_0
call-signaling-port 5060
bind interface IF_IP_LAN router

service default
domain 192.168.0.2
defaultserver manual 192.168.0.2 loose-router
registration-lifetime 300
registration manual 192.168.0.2
user 10001 authenticate name 10001 password 10001 default register

gateway sip GW_SIP_0
no shutdown

gateway sip GW_SIP_1
call-signaling-port 5062
bind interface IF_IP_LAN router

service default
domain 192.168.0.2
defaultserver manual 192.168.0.2 loose-router
registration-lifetime 300
registration manual 192.168.0.2
user 10002 authenticate name 10002 password 10002 default register

gateway sip GW_SIP_1
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_WAN router
no shutdown

port ethernet 0 1
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port bri 0 0
clock auto
encapsulation q921

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface IF_ISDN_0 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface IF_ISDN_1 switch

port bri 0 1
no shutdown

port bri 0 2
clock auto
encapsulation q921

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net

port bri 0 2
shutdown

 

[luppazzo]

can anyone reply to the prevous post?

thank you