Alcatel Unleashed

Hi,
I have an Alcatel-Lucent OXO release 7.0 with 2 sip trunk licenses. I am integrating it with an ISP SIP line and configurations configured as usual. The ISP technical team from their end can see the incoming calls and outgoing calls but their is no codec negotiation and therefore the calls are dropping at some point. At the client premise their is a Cisco router between the ISP and the Alcatel-Lucent OXO. If I interlink the same OXO with Cisco UC or othre SIP box the connection works but does not work with ISP provider SIP line. What problem could I be experiencing since I have changed all possible codecs on the OXO? Assist please.

Logs, god sake, logs

That's where the problem is because their no logs at all.

Just need to be guided where the possible problem is in such a case.

Logs, please.
Connect a hub (not a switch) between OXO and the LAN. Connect you computer on the same hub and get Wireshark traces. Post them here.
Crystal balls are not available.

Greetings.

wagash wrote:That's where the problem is because their no logs at all.

really !

wagash wrote:Hi,
The ISP technical team from their end can see the incoming calls and outgoing calls but their is no codec negotiation and therefore the calls are dropping at some point. At the client premise their is a Cisco router between the ISP and the Alcatel-Lucent OXO. "

this router plays the role of SBC ? ANY NAT THERE.
sip log, wireshark and more specif details on topology can help

yes no more magic ball lol

Hi all, as requested see additional details with logs

Incoming calls ring but drops/cuts when picked.

PABX OXO IP: 10.3.0.10
Gateway 10.3.0.100
ISP signalling IP 196.201.216.210

Wireshack incoming trace
1 0.000000000 196.201.216.210 10.3.0.10 SIP/SDP 1019 Request: INVITE sip:+212@10.3.0.10;user=phone |
2 0.525973000 196.201.216.210 10.3.0.10 SIP/SDP 1019 Request: INVITE sip:+212@10.3.0.10;user=phone |
3 1.526362000 196.201.216.210 10.3.0.10 SIP/SDP 1019 Request: INVITE sip:+212@10.3.0.10;user=phone |
4 3.527428000 196.201.216.210 10.3.0.10 SIP/SDP 1019 Request: INVITE sip:+212@10.3.0.10;user=phone |
5 7.526982000 196.201.216.210 10.3.0.10 SIP/SDP 1019 Request: INVITE sip:+212@10.3.0.10;user=phone |
6 15.526578000 196.201.216.210 10.3.0.10 SIP/SDP 1019 Request: INVITE sip:+212@10.3.0.10;user=phone |

My router config is as below:

interface FastEthernet0/0
description SAF
ip address 197.248.166.2 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto

!
interface Vlan1
ip address 10.3.0.100 255.255.0.0
ip nat inside
ip virtual-reassembly
!
!

ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 197.248.166.1
!
no ip http server
no ip http secure-server
ip nat source static 10.3.0.100 interface Vlan1
ip nat inside source list 100 interface FastEthernet0/0 overload
ip nat inside source list 107 interface FastEthernet0/0 overload
ip nat inside source static 10.3.0.10 197.248.166.2
!

access-list 107 permit tcp 10.3.0.0 0.0.0.255 eq 5060 any eq 5060
access-list 107 permit udp 10.3.0.0 0.0.0.255 eq 5060 any eq 5060
access-list 107 permit udp 10.3.0.0 0.0.0.255 range 1000 65530 any
access-list 107 permit udp any any
access-list 107 permit tcp any any
access-list 107 permit icmp any any
access-list 107 permit ip any any
access-list 107 permit ip 10.3.0.0 0.0.0.255 any
!

and below is trace/logs from our ISP end.

INVITE (Gateway to SIP PABX)

INVITE sip:+212@10.66.56.186;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.66.52.2:5071;branch=z9hG4bKiu8e9hhah8j8d8wdssicjacjs;X-DispMsg=1406
Route: <sip:10.66.56.186:5060;transport=udp;lr>
Call-ID: jaddyiauvv8u8bbywdeabahjbuaby7bv@10.18.5.64
From: "722000000"<sip:722000000@10.66.52.2;transport=udp;user=phone>;tag=89vjbb7w-CC-1024
To: "212"<sip:+212@10.66.56.186;transport=udp;user=phone>
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:10.66.52.2:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:722000000>
Supported: 100rel,timer,histinfo
Min-SE: 90
Session-Expires: 1800;refresher=uac
P-Early-Media: supported
Content-Length: 227
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1108924667 1108924668 IN IP4 10.66.52.2
s=SipCall
c=IN IP4 10.66.53.30
t=0 0
m=audio 24676 RTP/AVP 8 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5

100 ( From PABX)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.66.52.2:5071;branch=z9hG4bKiu8e9hhah8j8d8wdssicjacjs;X-DispMsg=1406
Call-ID: jaddyiauvv8u8bbywdeabahjbuaby7bv@10.18.5.64
From: "722000000"<sip:722000000@10.66.52.2;transport=udp;user=phone>;tag=89vjbb7w-CC-1024
To: "212"<sip:+212@10.66.56.186;transport=udp;user=phone>
CSeq: 1 INVITE
Content-Length: 0

180 (From PABX)

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.66.52.2:5071;branch=z9hG4bKiu8e9hhah8j8d8wdssicjacjs;X-DispMsg=1406
Record-Route: <sip:10.66.56.186:5060;transport=udp;lr>
Call-ID: jaddyiauvv8u8bbywdeabahjbuaby7bv@10.18.5.64
From: "722000000"<sip:722000000@10.66.52.2;transport=udp;user=phone>;tag=89vjbb7w-CC-1024
To: "212"<sip:+212@10.66.56.186;transport=udp;user=phone>;tag=sbc04071d2f9bad0788fcb5632a1af44f55d981
CSeq: 1 INVITE
Contact: "HOPE MASEKI"<sip:+254709962212@10.66.56.186:5060;user=phone>
User-Agent: OxO_GW_710/022.001
P-Asserted-Identity: "HOPE MASEKI"<sip:+254709962212@197.248.166.2;user=phone>
Content-Length: 0

488 (From PABX)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.66.52.2:5071;branch=z9hG4bKiu8e9hhah8j8d8wdssicjacjs;X-DispMsg=1406
Record-Route: <sip:10.66.56.186:5060;transport=udp;lr>
Call-ID: jaddyiauvv8u8bbywdeabahjbuaby7bv@10.18.5.64
From: "722000000"<sip:722000000@10.66.52.2;transport=udp;user=phone>;tag=89vjbb7w-CC-1024
To: "212"<sip:+212@10.66.56.186;transport=udp;user=phone>;tag=sbc04071d2f9bad0788fcb5632a1af44f55d981
CSeq: 1 INVITE

Hi

OXO is sending 488 Not acceptable Here.

Check your codec configuration in OXO side.
when ISP sends the INVITE, it is sending the codec negotiation:

a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000

OXO needs to match with ISP codecs.

Best regards.

Thanks for the feedback. Our ISP has set codec as G711. On the OXO I have tried all options on ARS prefixes section G711_10, G711_20, G711_30 and G711_60 but no change still. Am i changing the codec in the right section/area?