* -> Nortel BCM sip problem

Asterisk Asterisk Support
1

Hello all,

I currently have a basic Asterisk box setup with a sip trunk to our Nortel BCM. Everything works great calling between the 2, but I would like to take it one step further. I gave Asterisk access to the T-1 in the BCM. I can dial out no problem and answer the call, but after about 30 seconds it drops the call and gives a busy signal. Sip debug below:

I feel like it’s soooo close!!
TIA

[code]<------------->
— (16 headers 6 lines) —
 – SIP/HoustonBCM-092ce518 is making progress passing it to SIP/7001-092c5648
sterisk*CLI>
<— SIP read from UDP://10.11.11.200:5060 —>
SIP/2.0 183 Session Progress
From: "HoustonAsterisk"sip:7001@10.11.11.135;tag=as678499b2
To: sip:9XXXXXXXXXX@10.11.11.200;tag=40d496b0-c80b0b0a-13c4-4831f9b0-37e62dc4-4831f9b0
Call-ID: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.11.11.135:5060;rport=5060;branch=z9hG4bK4b96ab43
Supported: 100rel,sipvc
Require: 100rel
User-Agent: Nortel Networks BCM VoIP Gateway release_41 version_41.500.0.33
Privacy: None
x-nt-corr-id: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
RSeq: 534
Contact: sip:9XXXXXXXXXX@10.11.11.200
Allow: INVITE,UPDATE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK
Content-Type: application/SDP
Content-Length: 112

v=0
o=- 1211234736 1211234736 IN IP4 10.11.11.200
s=-
t=0 0
m=audio 28502 RTP/AVP 0
c=IN IP4 10.11.11.200

<------------->
— (16 headers 6 lines) —
 – SIP/HoustonBCM-092ce518 is making progress passing it to SIP/7001-092c5648
sterisk*CLI>
<— SIP read from UDP://10.11.11.200:5060 —>
SIP/2.0 183 Session Progress
From: "HoustonAsterisk"sip:7001@10.11.11.135;tag=as678499b2
To: sip:9XXXXXXXXXX@10.11.11.200;tag=40d496b0-c80b0b0a-13c4-4831f9b0-37e62dc4-4831f9b0
Call-ID: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.11.11.135:5060;rport=5060;branch=z9hG4bK4b96ab43
Supported: 100rel,sipvc
Require: 100rel
User-Agent: Nortel Networks BCM VoIP Gateway release_41 version_41.500.0.33
Privacy: None
x-nt-corr-id: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
RSeq: 534
Contact: sip:9XXXXXXXXXX@10.11.11.200
Allow: INVITE,UPDATE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK
Content-Type: application/SDP
Content-Length: 112

v=0
o=- 1211234736 1211234736 IN IP4 10.11.11.200
s=-
t=0 0
m=audio 28502 RTP/AVP 0
c=IN IP4 10.11.11.200

<------------->
— (16 headers 6 lines) —
 – SIP/HoustonBCM-092ce518 is making progress passing it to SIP/7001-092c5648
sterisk*CLI>
<— SIP read from UDP://10.11.11.200:5060 —>
SIP/2.0 183 Session Progress
From: "HoustonAsterisk"sip:7001@10.11.11.135;tag=as678499b2
To: sip:9XXXXXXXXXX@10.11.11.200;tag=40d496b0-c80b0b0a-13c4-4831f9b0-37e62dc4-4831f9b0
Call-ID: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.11.11.135:5060;rport=5060;branch=z9hG4bK4b96ab43
Supported: 100rel,sipvc
Require: 100rel
User-Agent: Nortel Networks BCM VoIP Gateway release_41 version_41.500.0.33
Privacy: None
x-nt-corr-id: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
RSeq: 534
Contact: sip:9XXXXXXXXXX@10.11.11.200
Allow: INVITE,UPDATE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK
Content-Type: application/SDP
Content-Length: 112

v=0
o=- 1211234736 1211234736 IN IP4 10.11.11.200
s=-
t=0 0
m=audio 28502 RTP/AVP 0
c=IN IP4 10.11.11.200

<------------->
— (16 headers 6 lines) —
 – SIP/HoustonBCM-092ce518 is making progress passing it to SIP/7001-092c5648
sterisk*CLI>
<— SIP read from UDP://10.11.11.200:5060 —>
SIP/2.0 500 Server Internal Error
From: "HoustonAsterisk"sip:7001@10.11.11.135;tag=as678499b2
To: sip:9XXXXXXXXXX@10.11.11.200;tag=40d496b0-c80b0b0a-13c4-4831f9b0-37e62dc4-4831f9b0
Call-ID: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.11.11.135:5060;rport=5060;branch=z9hG4bK4b96ab43
Supported: 100rel,sipvc
User-Agent: Nortel Networks BCM VoIP Gateway release_41 version_41.500.0.33
Privacy: None
x-nt-corr-id: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
Allow: INVITE,UPDATE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK
Content-Length: 0

<------------->
— (12 headers 0 lines) —
 – Got SIP response 500 “Server Internal Error” back from 10.11.11.200
Transmitting (no NAT) to 10.11.11.200:5060:
ACK sip:9XXXXXXXXXX@10.11.11.200 SIP/2.0
Via: SIP/2.0/UDP 10.11.11.135:5060;branch=z9hG4bK4b96ab43;rport
Max-Forwards: 70
From: “HoustonAsterisk” sip:7001@10.11.11.135;tag=as678499b2
To: sip:9XXXXXXXXXX@10.11.11.200;tag=40d496b0-c80b0b0a-13c4-4831f9b0-37e62dc4-4831f9b0
Contact: sip:7001@10.11.11.135
Call-ID: 6dd65ed1159f509171ab09bf37010917@10.11.11.135
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta9
Content-Length: 0

 – SIP/HoustonBCM-092ce518 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
 – Auto fallthrough, channel ‘SIP/7001-092c5648’ status is 'CONGESTION’
Really destroying SIP dialog ‘6dd65ed1159f509171ab09bf37010917@10.11.11.135’ Method: INVITE
asterisk*CLI>
[/code]

2

Dear cb900f
can you send to me a guide to how to do that
i want to connect asterisk system with my BCM 400

Thanks alot

3

Hi Memmy,

I don’t have a guide, but the BCM 400 4.0 apparently supports standard SIP protocols. In Asterisk I have an entry in sip.conf as:

[BCM]
type=peer
host=
canreinvite=no

Then on the BCM I just created a standard VoIP trunk to the Asterisk with SIP type of generic. You will also need keycodes for the BCM to allow VoIP trunks. You will also need to create a dialing plan on both sides (Asterisk and BCM).

Good luck!

BTW I am also on asterisk 1.4.20.1 as I had many SIP problems with 1.6 at the time.

4

Dear cb900f…
Thanks alot for your quick response and your cooperation
i try that and working for me [ Thanks :smile: ]

but i have a little problem
when i make that steps for the bcm in remote site doesn’t have the same IPs the asterisk didn’t connect to the BCM in remote site, when the asterisk server can ping to the BCM successfuly.

why ???

Thanks alot and have a nice day

5

Hi Memmy,

I’m a little confused. The asterisk does not have to be in the same subnet as the BCM, we currently have about 6 BCM’s and the asterisk can communicate no problem via VPN’s. Could you try to explain the problem a little better? Thanks!

6

Dear cb900f

that’s the problem :confused:

i have a BCM in my site, the asterisk connects to it with no problems. but the BCMs in the remote site the asterisk cannot connect to them :question:

the remote sites connected together via leased lines and cisco routers…

may be this is a network problem, i will try to solve it and if you have any new idea kindly inform me

Thanks alot my friend

7

It does sound like a network problem. You can try a trace route to possibly track down the issue. But if you cannot ping the remote BCM’s from the asterisk then it’s definately a network problem. Wish I could help more.