voip-voip gateway using Asterisk and 2 Broadvoice lines

[rdmoore]

I am an Asterisk beginner. I have been using a Sipura SPA-3000 as a voip-voip gateway using 2 Broadvoice lines. I wish to use Asterisk to replace it. I took the SPA-3000 off-line. I have Asterisk version 1.0.9 installed and configured, but I can't get it to work. I would appreciate any assistance.

The two numbers register with no problem, but when I call the entry number I get a fast busy. I am sure I am missing something. Here are my config files:

sip.conf file:

[general]
pedantic=no
port=5060
context=incoming
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm

;-----------------------------------------------------------------------------------

register => 330XXXXX93@sip.broadvoice.com:secret...broadvoice.com/330XXXXX93
register => 330XXXXX77@sip.broadvoice.com:secret...broadvoice.com/330XXXXX77

;-----------------------------------------------------------------------------------

[330XXXXX93] ; Entry Number
type=peer
;user=phone
host=sip.broadvoice.com
fromuser=330XXXXX93
secret=secret
username=330XXXXX93
insecure=very
context=in-sip
authname=330XXXXX93
dtmfmode=inband
dtmf=inband
canreinvite=no
nat=yes

[330XXXXX77] ; Termination Number
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=330XXXXX77
secret=secret
username=330XXXXX77
insecure=very
context=outgoing
authname=330XXXXX77
dtmfmode=inband
dtmf=inband
canreinvite=no
nat=yes


extensions.conf file:

[general]
static=yes
writeprotect=no

;-----------------------------------------------------------------------------------

[outgoing]
exten=>_1NXXNXXXXXX,1,dial(SIP/${EXTEN}@330XXXXX77,30)
exten=>_1NXXNXXXXXX,2,congestion()
exten=>_1NXXNXXXXXX,102,busy()

[incoming]
exten=>s,1,Wait,1
exten=>s,2,Answer
exten=>s,3,responsetimeout(10)
exten=>s,4,DigitTimeout(3)
exten=>s,5,ResetCDR(w)
exten=>s,6,DISA(no-password|outgoing)
;exten=>s,6,DISA(/etc/asterisk/disa.conf)
exten=>s,7,Congestion

[in-sip]
exten=>330XXXXX93,1,goto(incoming,s,1)
exten=>330XXXXX77,1,goto(outgoing,s,1)


Thanks in advance your help.

[thameema]

what are you seeing in the asterisk log or CLI? Is it only problem with broadvoice or is it problem with all other lines?

[mberlant]

I recommend that you configure your SPA-3000 for attachment to your Asterisk. If you use the Voxilla Wizard it will set up an ordinary extension for Line 1 and a Dial-in/Dial-out extension for PSTN Line. This will be a good troubleshooting tool. You may also wish to set up a PC with X-Lite so you can have a second extension for testing and for use.

Also, I notice that your outbound contexts are a little bit different from my successful ones. You may want to adjust yours to mimic mine.

[330XXXXX93]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=330XXXXX93
secret=a1b2c3d4e5
authuser=330XXXXX93
username=330XXXXX93
canreinvite=no
insecure=very
disallow=all
allow=ulaw
allow=alaw
auth=md5,plaintext
dtmfmode=inband
qualify=yes
nat=yes

[rdmoore]

Thank you for your responses.

tharneema,

When I run sip debug I get the following messages:

With no call activity the following message shows every few seconds
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.230;5060;received=24.123.252.210;branch= z9hG4bK58073b35;rport=63416
From: <sip:330XXXXX93@sip.broadvoice.com>;tag=as31736b 6e
To: <sip:330XXXXX93@sip.broadvoice.com>
Call-ID: 2b0a94e16a3e085d545f2282135032cf@127.0.0.1
CSeq: 192 REGISTER
Contact: sip:330XXXXX93@192.168.1.230>;expires=20

7 headers, 0 lines
Sep 13 11:08:11 NOTICE[1146]chan_sip.c:6831 handle_response:Outbound Registration:Expiry for sip.broadvoice.com is 20 sec (Scheduling reregistration in 15999 ms)
Destroying call '2b0a94e16a3e085d545f2282135032cf@127.0.0.1'


When I call the Entry number I get the following message:
Sip read:
ACK sip:330XXXXX93@192.168.1.230 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3a98of207odgaag2f 500.1sr
From: "CID NAME"<sip:330XXXXX23@147.135.0.129;user=phone>;tag =SD2gi9001-1051128971-1126547148669
To: "CID NAME"<sip:330XXXXX93@sip.broadvoice.com;user=phone >;tag=as22463909
Call-ID: SD2gi9001-7562acf1569dd02df5cfc3504399392e-js11002
CSeq: 632858559 ACK

6 headers, 0 lines
Destroying call 'SD2gi9001-7562acf1569dd02df5cfc3504399392e-js11002


mberlant,

I used your example for my contexts except for the line "auth=md5,plaintext". I still have the same problem.
I noticed you don't have a "context=" line. Is it not required?
I will be taking your suggestions and using the SPA-3000 and a PC with X-Lite for troubleshooting. In the mean time, any more suggestions?

[mberlant]

With Asterisk there are several ways to accomplish just about any task at hand. The way I understand it you don't need a context= line if you have a /12223334444 tag at the end of the register= line, because the incoming call will be automatically thrown over to the [in-sip] context unless overridden with a context= line.

[rdmoore]

Thanks for the answer. Are there any other config files I should be looking at besides sip.conf and extensions.conf?

[mberlant]

For SIP connections, I don't think so. The basic architecture is that all work and decisionmaking occurs in extensions.conf and its add-ins. Major features have their own configuration files: sip.conf, iax2.conf, voicemail.conf, disa.conf, etc. Which ones you care about depend on what features you need.

[rdmoore]

There must be something wrong with either sip.conf, extensions.conf, or my network.
Since both BV lines worked just fine with the SPA-3000, and they registered OK with Asterisk, I assumed that my network configuration is fine. That is why I was concentrating on the config files. Also, as you can see from the sip debug messages above, the asterisk system sees the incoming call, it just doesn't do anything with it. Is it possible that I need to change something in my router? I have the IP address of the asterisk server in the DMZ.

[rdmoore]

I have made some progress. In sip.conf, I changed "context=" line to "context=in-sip" for [general], [330XXXXX93], and [330XXXXX77]. Now I get a dialtone. When I dial a number, after a few seconds I get a busy signal. Do I need to change something in the [outgoing] context in extensions.conf?

[mberlant]

You will progress more quickly if you set up local extensions and test each path independently before trying to test them back-to-back.