[thameema]

I have two bv accounts with my asterisk. I don't have problem making outgoing calls but incoming calls are problem. I can only get incoming calls from only one account and not both. I tried all possibilities and found that it should be asterisk issue handling two bv registrations. If anyone succeeded with this kind of configuration please post your configuration.

Thanks.

[mberlant]

I have had more than one BV account running through my Asterisk for several months now, and they all coexist without a problem.

The two halves of the solution, making calls and receiving them, need to be addressed independently.

You say you are able to make calls. Could you confirm that you have appropriate control over which account is used for each call? That is, can you reliably select which account is used?

For receiving calls you need to have a separate register= line in sip.conf and a corresponding exten line in [in-sip] in extensions.conf which points to a separate context to handle each line. My register= lines look like this (sanitized, of course):

register=3115552368@sip.broadvoice.com:a1b2c3d4e5:3115552368@sip.broadvoice.com/3115552368
register=2135550196@sip.broadvoice.com:f6g7h8i9j0:2135550196@sip.broadvoice.com/2135550196
register=9784187300@sip.broadvoice.com:k1m2n3o4p5:9784187300@sip.broadvoice.com/9784187300

Make sure that the routing tail (the bit at the end after the /) has the ten-digit primary phone number for the appropriate account.

Then, in extensions.conf:

[in-sip]
exten => 3115552368,1,Goto(from-broadvoice-2368,s,1)
exten => 2135550196,1,Goto(from-broadvoice-0196,s,1)
exten => 9784187300,1,Goto(from-broadvoice-7300,s,1)

These lines take the "handoff" from the register= lines in sip.conf.

[from-broadvoice-2368]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Dial(SIP/101&SIP/133,30,t)
exten => s,4,Voicemail(u101)
exten => s,5,Hangup
exten => s,104,Voicemail(u101)
;
[from-broadvoice-0196]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Dial(SIP/122&SIP/133,30,t)
exten => s,4,Voicemail(u122)
exten => s,5,Hangup
exten => s,104,Voicemail(u122)
;
[from-broadvoice-7300]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Dial(SIP/105&SIP/133,30,t)
exten => s,4,Voicemail(u105)
exten => s,5,Hangup
exten => s,104,Voicemail(u105)

I hope this example helps.

[thameema]

Thanks mberlent. I followed your instructions and it worked 90%. The calls are coming thru and ringing the extensins properly. But the only thing i noticed is, if i call my first bv number its going thru properly. If i call the second number its ringing the extensions but in the asterisk logs, i m seeing my first number in the records..

my first bv number is 925xxxx042 and my second number is 614xxxx309.

Here is the snippet for when i call my first bv number which looks good.

   -- Executing Goto("SIP/925xxxx042-8bc8", "call-home-1|s|1") in new stack
    -- Goto (call-home-1,s,1)
    -- Executing Answer("SIP/925xxxx042-8bc8", "") in new stack
    -- Executing Dial("SIP/925xxxx042-8bc8", "SIP/200|60|t") in new stack
    -- Called 200
    -- SIP/200-c83b is ringing
    -- SIP/200-c83b answered SIP/925xxxx042-8bc8

here is the log snippet when i called my second number.

    -- Executing Goto("SIP/925xxxx042-3594", "call-home-2|s|1") in new stack
    -- Goto (call-home-2,s,1)
    -- Executing Answer("SIP/925xxxx042-3594", "") in new stack
    -- Executing Dial("SIP/925xxxx042-3594", "SIP/301|60|t") in new stack

In my sip.conf i have this:

;bv accounts
register=>614xxxx309@sip.broadvoice.com:<secret>:614xxxx309@sip.broadvoice.com/614xxxx309
register=>925xxxx042@sip.broadvoice.com:<secret>:925xxxx042@sip.broadvoice.com/925xxxx042

my extensions.conf have this entry:

[from-broadvoice]
exten => 614xxxx309,1,Goto(call-home-2,s,1)
exten => 925xxxx042,1,Goto(call-home-1,s,1)

I also have the entries for call-home-1 and 2 in my extensions.conf to dial the corresponding extensions.

Do you have any clue about why my first bv number is appearing in call log when i call the second bv number.

The outgoing is fine because i can see two different entries.

Thanks in advance.

[mberlant]

Yes, I have observed this phenomenon with regard to all services on my Asterisk that have more than one account associated with them. This includes BroadVoice, StanaPhone, FWD, SIPphone and a couple of others. Call processing is correct, but the log and channel info reports show the first one registered, regardless of which one was actually in use. What I do about this is to prepend the channel name onto the Calling Name ID. So, "John Smith" becomes "BV2368-John Smith".

[rjarow]

So albeit, I'm VERY new to this software. I am pretty good with linux and know my away around hardware etc.

That being said, I setup a home *@h box, and it's working great, 2 trunks voipjet outgoing, broadvoice incoming.

Now, I am at work, attempting to setup a single trunk setup, using broadvoice unlimited package.

I am able to do outgoing calls fine, but incoming is a no go.

I've practically mirrored my good, home workign config on this server (simply swapping in diff't phone # and password) and still, when I call I get "This line is busy your call cannot be completed..etc"

I do see a quick entry of the call in asterisk console w/ sip debug on but that's it.

I've followed the tutorial @ chayden.net to get both servers setup, and has worked great, just wondering what may be different here on this setup causing problems.

I can and will post config entries if needed, just don't want to flood the original post. Any and all help will be appreciated.

Sidenote: this is a great community, and glad I found it.

[rjarow]

Re: My post being moved here

I'm not sure this should have been moved.

As my setups are independent of eachother, not using the same broadvoice account (2 sep accts, and 2 sep servers)

However, any help would be appreciated.

[mberlant]

The advice in this thread applies directly to your problem. You need to make sure that your sip.conf and extensions.conf segments perform all of the functions described here.

[rjarow]

Well I don't have lines similar to those above on my working *@home server, so I'm not sure that's exactly it. I'm only trying to get the same deal working, just with a different account and different server.

Here is my sip_additional.conf snippet of the trunks

my extensions.conf snippet

register=2136341497@sip.broadvoice.com:<removed>:2136341497@sip.broadvoice.com/0002136341497




[2136341497]
username=2136341497
user=phone
type=user
secret=<removed for security reasons>
qualify=100
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn
fromuser=2136341497
authname=2136341497
callerid=2136341497

[broadvoice]
username=2136341497
user=phone
type=peer
secret=<removed for security reasons>
insecure=very
host=sip.broadvoice.com
fromuser=2136341497
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn
canreinvite=no
authname=2136341497

my sip.conf top

[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
pedantic=no
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

now extensions.conf snippet

[from-sip-external]

;give external sip users congestion and hangup
;exten => _.,1,AbsoluteTimeout(15)
;exten => _.,2,Congestion
;exten => _.,3,Hangup

exten => _.,1,Wait(1)
exten => _.,2,Goto(from-pstn,s,1)

Hopefully this helps?
NOTE: I removed passwords, in actual conf files the authpassword is present.

[rjarow]

Seems like we are registering ok, no?

Reliably Transmitting:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK4d5f4b53
From: <sip:2136341497@sip.broadvoice.com>;tag=as5af9c8 c2
To: <sip:2136341497@sip.broadvoice.com>
Call-ID: 6c104773566bc9e436687c572e6f8134@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:0002136341497@xx.xx.xx.xx>
Event: registration
Content-Length: 0

(no NAT) to 147.135.8.128:5060
asterisk1*CLI>

Sip read:
SIP/2.0 200 OK
Call-ID: 6c104773566bc9e436687c572e6f8134@127.0.0.1
CSeq: 102 REGISTER
From: <sip:2136341497@sip.broadvoice.com>;tag=as5af9c8 c2
To: <sip:2136341497@sip.broadvoice.com>
Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK4d5f4b53
Contact: <sip:2136341497@xx.xx.xx.xx>
Expires: 672
Event: registration
User-Agent: Asterisk PBX
Content-Length: 0


---

I removed the IP because it is an external box, no offense to you all

[rjarow]

We do get some sort of response when someone does call.

Sip read:
ACK sip:2136341497@##.##.##.##:5060 SIP/2.0
Call-ID: ff034f-d@147.135.8.128
CSeq: 1 ACK
From: "XXX XXXXX "<sip:XXXXXXXXXX@##.##.##.##;user=phone>;tag=e gij
To: "XXXX XXXXXX"<sip:XXXXXXXXXX@##.##.##.##;user=phone>;tag =as09f1f352
Via: SIP/2.0/UDP 147.135.8.128:5060;received=72.34.43.48
Content-Length: 0


7 headers, 0 lines
Destroying call 'ff034f-d@147.135.8.128'