[quibbly]

I use Teliax as one of my voip providers on my asterisk@home server. This works like a charm for both inbound and outbound calls.

Just added Broadvoice to the picture and I'm able to make outbound calls, but the inbound calls do not work, just a fast busy. The log file shows:

816278#### is not a local user

816278#### is my Broadvoice number

This is my AMP outgoing setting:
authname=816278####
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=816278####
host=sip.broadvoice.com
insecure=very
nat=yes
secret=xd#######
type=peer
user=phone
username=816278####

This is my AMP outbound Setting:

context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=xd######
type=user
user=816278####
username=816278####

Please help!

thanks

[mberlant]

That message indicates that your Asterisk is successfully receiving the call from BroadVoice, but extensions.conf doesn't know what to do with it.

Now, I can't speak to how AMP has got your Asterisk set up, but I can tell you that your problem is that your

exten => 816278####,1,Dial(SIP/200)

or similar command line is either in the wrong [context] or is missing altogether. Near the top of your sip.conf file is a "register =>" line with your BroadVoice information. It likely ends with /816278####. Trace up from that line for a "context=..." line. When a call comes in from BV, sip.conf puts it in a queue with that context name for extensions.conf to pick up and handle. Now, go to extensions.conf and look for the exten line similar to the one above and make sure that extensions.conf is looking to pick it up from the same named [context] that sip.conf put it in.

Good luck.

[rdbeamer]

the broadvoice registry string is located in the sip_additional.conf file on my box and looks like this:
register=317xxxxxxx:xxxxxxxxxx@sip.broadvoice.com
Know on one of the earlier builds of asterisk, it didn't configure it correctly, and it took awhile to figure that one little thing out.
Good luck

[quibbly]

Asterisk and Asterisk@home configure the files a little different. The above info is not in the sip.conf file
I'm using Asterisk@home and I'm using Teliax for one voip provider and it works like a charm. I can send and receive calls with no problem
Setup Broadvoice and I'm able to dial out via my DID setup (8|*), but inbound does not work. It shows to ring on my Teliax-Inbound. Could it be a problem having "context=from-pstn " configured in both Telix and Broadvoice dialout? Telix has a setting for inbound calls via asterisk@home while Broadvoice only has configuration for outbound setting.

rdbeamer, I have the register line setup. That allows me to call out.

If you guys use asterisk@home, let me know so I can use your configuration. I've tried everything I could think off and 2 hrs sleep in 48hrs trying to get this thing to work. I'm probably going to dump asterisk@home and do a full asterisk install so I can actually configure via sip.conf.

[mberlant]

Whether you or AMP configure Asterisk's .conf files, the Asterisk engine is the same.

The actual function of the register line is to handle inbound calls. Some providers, like BV, require registration in order to make outbound calls, but there is no direct outbound functionality in the register line. As I said above, the /###### at the end of the register line determines how the inbound call is handled and to where it is initially routed. If you are missing this bit of information, Asterisk will treat the incoming call as uninvited and will handle it according to its "default" inbound handling scheme. This may be why quibbly sees his inbound BV calls identified as being via Teliax.

If you want to treat your inbound BV calls specially, you have several options, and all involve putting /#### behind the appropriate register command. If you can get AMP to do this for you, more power to you, but it must be done regardless in order to succeed. Basically, whatever context is specified in the most recent "context=" line above the register line, that is the [context] that the matching "exten => ####" line must be beneath in your extensions.conf (or its additional) file.

The convention I use is to put the phone number, preceded by three zeroes, as this "routing" number. My register line is "register=3115552368:bvpasswd:3115552368@sip.broad voice.com/0003115552368". In my sip.conf (which may be in your sip_additional.conf, or similar) file, this is underneath a "context=in-sip" line. This means that when a call comes in it is "handed over the fence" as "in-sip/0003115552368".

My extensions.conf (your extensions_additional.conf file?) file is constantly watching this queue for activity. When it sees "in-sip/0003115552368" appear in the queue it looks in its tables for the [in-sip] context and, if it finds it, then looks for "exten => 0003115552368,1,........." within that context. Upon finding that exten line it executes whatever command is in there, like "Dial(SIP/2200)", if I want all BV calls to ring Extension 2200.

It doesn't really matter where within your sip.conf environment (the main file or its additions) the register line is nor where within your extensions.conf environment the particular exten line is. The only thing that really matters is that sip.conf sends it over the fence in such a manner that extensions.conf can catch it and run with it, if you will forgive a sports analogy.

I hope that someone out there has figured out how to make AMP accomplish this. All I can do it explain what must ultimately happen within Asterisk.

[quibbly]

mberlant,

Thank you for your help. What you explained makes perfect sense. I will be reformatting my system this evening and installing full asterisk, not the A@H version. This will allow me to follow advice from people who use this daily. A@H is just a version from my understanding, to make it easier, but it can't be updated.

I have the Grandstream GXP2000 and Wildcard TDM400P with FXO and FXS arriving today. Should have fun playing with this and having a full Asterisk system to work with.

Thank you guys for your help, and with your advice, I will have Broadvoice working this evening.

Thanks,

Quibbly

[mberlant]

Quibbly,

It is up to you, of course, but you need not replace your Asterisk@Home with vanilla Asterisk to take control of this piece of the puzzle. I, for example, have installed Asterisk@Home because I was having trouble getting a full Linux package to install cleanly on my Pentium MMX machine. I installed Asterisk@Home and then copied in my own .conf files and the machine came to life.

I don't know what your Linux skill level is, but whichever path you choose you will need to know enough about moving files around and editing them. Good luck.

Michael

[quibbly]

Originally Posted by mberlant

Quibbly,

It is up to you, of course, but you need not replace your Asterisk@Home with vanilla Asterisk to take control of this piece of the puzzle. I, for example, have installed Asterisk@Home because I was having trouble getting a full Linux package to install cleanly on my Pentium MMX machine. I installed Asterisk@Home and then copied in my own .conf files and the machine came to life.

I don't know what your Linux skill level is, but whichever path you choose you will need to know enough about moving files around and editing them. Good luck.

Michael

I am having more fun with standard * instead of the a@h. I'm actually pretty fluent with linux due to my job. I'm Director of IT for large medical group and we use multiple SCO Unix servers (yep..SCO). I cut my teeth on this well before NT4.0, 2000, 2003. About to implement a massive Blade server setup connected to CX300 FibreSan devices. 3 of the blades are RH Enterprise for its load balance (Tomcat frontend of application).

Just a little background about myself. I do some programming in ASP, jscript, .net, etc. but now I'm getting more involved in linux software development. Have to say, its more fun!

Enjoy,

Quibbly

[isepic]

Originally Posted by quibbly

I use Teliax as one of my voip providers on my asterisk@home server. This works like a charm for both inbound and outbound calls.

Just added Broadvoice to the picture and I'm able to make outbound calls, but the inbound calls do not work, just a fast busy. The log file shows:


thanks

just one thing to check is your FW port, if you have a firewall, be sure to point the incoming port to the proper internal server IP.

I kept switching around between boxes and forgot, so I got nothing but fast busy, and this was the cause... I had forgottento change the inbound port on my linksys to point to the new maching on the IAX port.

[billn]

I have a similar problem to the one described above. Broadvoice outbound calls work perfectly. But on inbound, I only get a busy signal.

Here are the applicable bits of config:

sip.conf

context=from-broadvoice
register => 7075044511@sip.broadvoice.com:XDxxxx...broadvoice.com/0007075044511


[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=7075044511
secret=XDxxxxxxxxxx
username=7075044511
insecure=very
context=from-broadvoice
authname=7075044511
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=yes


[101]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=101
secret=xxxxxxxx
;nat=yes=1234 ; X-Lite is behind a NAT routertension 1234
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailbo
context=novatrope



and from extensions.conf

[novatrope]
exten => 101,1,Dial(SIP/101,,rm)
ignorepat => 9
exten => _91NXXNXXXXXX, 1, dial(SIP/${EXTEN:1}@sip.broadvoice.com,30)
exten => _91NXXNXXXXXX, 2, congestion()
exten => _91NXXNXXXXXX, 102, busy()

[from-broadvoice]
exten => 0007075044511,1,Dial(SIP/101)

The logs show the call coming in and being routed to 0007075044511@ip

Any ideas would be appreciated.