Settings for Broadvoice ?
Re: Settings for Broadvoice ?
Tue, 06/06/2006 - 7:44am
Found this info on the quick install guide, hope this is of some help to you.
If they are using broadvoice make sure you let them know not to use broadvoice help site for asterisk. Here is a sip.config i had to use.After many night of getting pissy with the server i found this and it works for the incoming and outgoing.
Mark
[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
allow=ulaw
context=default ; Added to Test
srvlookup=yes ; Added to Test
pedantic=no ; Added to Test
;context=inbound ; Send unknown SIP callers to this context
;context=from-sip-external ; Send unknown SIP callers to this context
callerid=Unknown
register => [THERE PHONE NUMBER]@sip.broadvoice.com:[PASSWORD]:[THERE PHONE NUMBER]@sip.broadvoice.com
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
[sip.broadvoice.com]
callerid=[THERE PHONE NUMBER]
context=from-pstn
dtmfmode=rfc2283
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
secret=[PASSWORD]
type=user
user=[THERE PHONE NUMBER]
username=[THERE PHONE NUMBER]
[outgoing]
authname=[THERE PHONE NUMBER]
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=[THERE PHONE NUMBER]
host=sip.broadvoice.com
insecure=very
nat=no
secret=[PASSWORD]
type=peer
user=phone
username=[THERE PHONE NUMBER]
Re: Settings for Broadvoice ?
Thu, 06/08/2006 - 7:49pm
Hi Guys,
I'm a newb, and was wondering if you could be more detailed in where these settings should be put. Doesn't look like it all goes in sip.conf. I'm guessing the last block goes in to extensions.conf?
I downloaded trixbox and burned the iso image. I installed the software onto a suitable machine but not too clear on what to do from this point. The documentation I find seem to point all over the place and no clear cut way to setup for broadvoice. I just want to get 6 soft sip phones to be able to dial out and recieve calls through this PBX that was supposed to be so easy to setup.
You are right, Broadvoice is useless support is useless.
Please help.
Re: Settings for Broadvoice ?
Sat, 06/10/2006 - 2:14pm
I have a broadvoice line for mostly incoming and backup outgoing:
These settings are from Asterisk@Home 1.5 but should work on trixbox (I'm just getting my trixbox server built and will cutover in a week or so):
Trunk SIP/sip.broadvoice.com
Outgoing Caller ID: XXXYYYZZZZ
Max channels: 2
Outgoing Settings:
Trunk name: sip.broadvoice.com
Peer details:
allow=ulaw
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=sip.broadvoice.com
fromuser=XXXYYYZZZZ
host=sip.broadvoice.com
insecure=very
nat=yes
secret=password
type=peer
username=XXXYYYZZZZ
Incoming settings: [nothing here]
User context and User details are blank
Register string:
XXXYYYZZZZ@sip.broadvoice.com:password:XXXYYYZZZ@sip.broadvoice.com/XXXYYYZZZZ
Where XXXYYYZZZZ is my 10 digit broadvoice number and password is my broadvoice SIP password.
Very importantly, choose the best IP address for sip.broadvoice.com and put an entry in your /etc/hosts file:
My /etc/hosts file has a primary and backup:
#dca and bos ~50ms 3/20/2005 ~20ms 4/17/06
#147.135.0.128 sip.broadvoice.com
#nyc ~14ms 4/17/06
147.135.20.128 sip.broadvoice.com
Then add DID/Incoming route for XXXYYYZZZZ to whatever extension or ringroup or whatever you desire.
Re: Settings for Broadvoice ?
Sat, 06/10/2006 - 5:20pm
See:
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
Peer settings take care of everything just fine.
Re: Settings for Broadvoice ?
Sat, 06/10/2006 - 5:59pm
heres what I found to work
peer
allow=ulaw
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=sip.broadvoice.com
fromuser=mynumber
host=sip.broadvoice.com
insecure=very
nat=yes
secret=password
type=peer
username=mynumber
User context
context=from-pstn
dtmf=rfc2833
dtmfmode=rfc2833
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=password
type=friend
user=mynumber
username=mynumber
mynumber@sip.broadvoice.com:password:mynumber@sip.broadvoice.com
Re: Settings for Broadvoice ?
Mon, 06/12/2006 - 10:34am
After long days and nights of probing the web this is what has worked for me. This is on Trixbox using FREE PBX Admin.
Outbound Caller ID:
Maximum Channels: <2>
Dial rules : <1|NXXNXXXXXX>
Trunk Name:
Peer Details:
authname=
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=
host=sip.broadvoice.com
insecure=very
qualify=yes
secret=
type=peer
user=phone
username=
INCOMING SETTINGS
User Context:
User Details:
authname=
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=
host=sip.broadvoice.com
insecure=very
secret=
type=user
user=phone
username=
Registation String:
Make sure you omit the <> signs and enter your variables. I have copied this directly from my working broadvoice trunk and removed my variables.
Re: Settings for Broadvoice ?
Fri, 06/23/2006 - 12:20pm
Cowboy--That problem could be coming from your dial rules. You'll need something like this NXXNXXXXXX entered for long distance calls. Also, check your outbound routing. In one place or the other Askerisk isn't sure which trunk to use based on the number you dialed. Play around with some different combinations and read the guides and you'll get it. At first the concept is hard to wrap your head around, but eventually you'll say "dooohhh!'
Re: Settings for Broadvoice ?
Fri, 06/23/2006 - 12:34pm
Thanks RBowles for helping me figure this out and understand it.
My experiance is coming from AAH 2.5, but should be the same for newer versions. Incoming settings are no longer used (see http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer )and were preventing me from getting calls on SIP trunks. Well, the calls were coming in, but under the wrong context (from-sip-external) which AAH/Trixbox sends to congestion/busy by default, and never rings an extension, As soon as I removed the trunk's Incoming Settings, everything worked as expected.
Re: Broadvoice Distinctive Ring
Mon, 06/26/2006 - 6:13pm
BraodVoice insists on directing all calls to your main number, so you can't use DID Numbers directly.
This what works for me.
1) Get BroadVoice working as a single line.
2) Add the following to the end of /etc/asterisk/extensions_custom.conf:-
[custom-bv-incoming]
exten => s,1,Noop(Alert-Info -> '${SIP_HEADER(Alert-Info)}')
exten => s,2,Gotoif($["${SIP_HEADER(Alert-Info)}" = "
exten => s,3,Gotoif($["${SIP_HEADER(Alert-Info)}" = "
exten => s,4,Goto(from-pstn,bv-main,2)
(That's meant to be be 4 lines in case it gets wrapped)
3) Edit the Inbound Route destination for your main BroadVoice number to be a custom app with the following value:-
custom-bv-incoming,s,1
4) Now tell Asterisk how you want calls to be processedby setting up Inbound Routes using the following as DID Numbers:-
bv-main
bv-dr3
bv-dr4
Freepbx will complain that non-numeric destinations is an advanced use. Just hit 'OK' to ignore the complaints.
This is for the default distinctive ring settings on BroadVoice.
Note you can change 'bv-dr3' and 'bv-dr4' to be the actual numbers. You can NOT use the actual number to replace 'bv-main'
Re: Settings for Broadvoice ?
Sun, 07/02/2006 - 8:21am
Well, I don't think that is the problem but thanks. I have my dial rules assigned and while I am not doing a round robin for outbound trunks and I have assign specific keys for outbound (i.e. 7| - broadvoice, 8| - aeris communications, 6| - dow networks) all outbound routes have dial rules and the trunks as well. Before I changed over to 1.1 I had both 7 & 8 working fine. One of my problems with broadvoice was in the context= statement. But now, some work, others dont. All incoming work fine, it is just with the outbound that I have the problem.
Aeris claims that there is a problem with the password when registering. They have recreated the problem but do not know how to solve it. While my password and register string are correct, when it gets to their switches (Cisco & Lucent) it is rejected. Aeris told me that Cisco & Lucent told them to buy new equipment or buy a software upgrade. Seems strange that this would appear from 1.0 to 1.1.
C
Re: Settings for Broadvoice ?
Fri, 07/14/2006 - 8:58pm
Please forgive me for my ignorance but I guess I am going to need a dummies guide approach on how to configure broadvoice with trixbox 1.1. I have been looking at all of these posts and they seem confusing as to how to properly configure within freepbx. Is it possible to do so? I have an all softphone network waiting for the broadvoice to work with it. Any/all help would be greatly appreciated.
UPDATE: Got outgoing to work but incoming is still a problem...
When I call my broadvoice number from my cell, I get "The number you have reached is not in service"...
Re: Settings for Broadvoice ?
Sat, 07/15/2006 - 6:30am
This is a working config. My only complaint with broadvoice is on the quality of the calls. It is my understanding that they will only support the ulaw codecs. On my other providers I am using GSM and the quality is much better. I have this setting set up in both sip_additional.con and in a file that I created manually "sip_custom.conf" I did this because I always have a working copy of my config in the event sip_additional.conf is changed by freepbx.
U=userid or phone number provided by Broadvoice
P=password provided by Broadvoice (this password is not the one used to access your web account, broadvoice will provide you with the sip information)
register=UUUUUUUUUU@sip.broadvoice.com:PPPPPPPPPP:UUUUUUUUUU@sip.broadvoice.com/200
[8177650032]
username=UUUUUUUUUU
user=UUUUUUUUUU
type=user
secret=PPPPPPPPPP
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
dtmf=rfc2833
disallow=all
context=from-pstn
canreinvite=yes
authname=UUUUUUUUUU
allow=g711u
allow=g711a
allow=ulaw
monitor=yes
rtpupdate=yes
[sip.broadvoice.com]
username=UUUUUUUUUU
type=peer
secret=PPPPPPPPPP
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=UUUUUUUUUU
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
disallow=all
context=from-pstn
canreinvite=yes
allow=g711u
allow=g711a
allow=ulaw
monitor=yes
rtpupdate=yes
•On the register line, the last item “/200” refers to the extension or number you want the calls pointed to. In my case I want the calls to be pointed to extension 200 on my system.
•Make sure that you place the ip address for sip.broadvoice.com in your /etc/hosts file.
Re: Settings for Broadvoice ?
Sat, 07/15/2006 - 2:16pm
I tried your settings but no luck, here is what is working for me for outgoing: (used freepbx to apply all settings below)
[sip.broadvoice.com]
authname=NNNNNNNNNN
canreinvite=no
context=from-pstn
dtmf=rfc2833
dtmfmode=rfc2833
fromdomain=sip.broadvoice.com
fromuser=NNNNNNNNNN
host=sip.broadvoice.com
insecure=very
nat=yes
qualify=yes
secret=PPPPPPPPPPP
type=peer
user=phone
username=NNNNNNNNNN
Here is my incoming settings:
[NNNNNNNNNN]
context=from-pstn
dtmf=rfc2833
dtmfmode=rfc2833
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=PPPPPPPPPP
type=user
user=NNNNNNNNNN
username=NNNNNNNNNN
Here is my registration string:
NNNNNNNNNN@sip.broadvoice.com:PPPPPPPPPPP:NNNNNNNNNN@sip.broadvoice.com
Here is my Inbound Route:
DID Number=
Caller ID Number=NNNNNNNNNN
Set Destination=Core:
*Everything else was left at default settings
Here is my extension 200 config:
(thought these values might have value in debugging)
dtmfmode=rfc2833
type=friend
context=from-internal
*Everything else was left at default settings
Debug Info from asterisk console:
<-- SIP read from 147.135.0.128:5060:
INVITE sip:s@192.168.1.224 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK4ael0810congvccdo7s0.1sr
From: "Somewhere TX"
To: "John Doe"
Call-ID: SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002
CSeq: 971341210 INVITE
Contact:
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel
Accept: multipart/mixed,application/sdp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 266
v=0
o=BroadWorks 277841056 1 IN IP4 147.135.0.128
s=-
c=IN IP4 147.135.0.128
t=0 0
m=audio 12326 RTP/AVP 0 8 2 18 96 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 G729AB/8000
a=rtpmap:97 iLBC/8000
--- (13 headers 12 lines)---
Using INVITE request as basis request - SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002
Sending to 147.135.0.128 : 5060 (non-NAT)
Found peer 'sip.broadvoice.com'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Peer audio RTP is at port 147.135.0.128:12326
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format G729AB
Found description format iLBC
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for s in from-pstn (domain 192.168.1.224)
list_route: hop:
Transmitting (NAT) to 147.135.0.128:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK4ael0810congvccdo7s0.1sr;received=147.135.0.128
From: "Somewhere TX"
To: "John Doe"
Call-ID: SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002
CSeq: 971341210 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0
---
-- Executing NoOp("SIP/NNNNNNNNNN-e8c4", "No DID or CID Match") in new stack
-- Executing Answer("SIP/NNNNNNNNNN-e8c4", "") in new stack
We're at 192.168.1.224 port 19986
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (NAT) to 147.135.0.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK4ael0810congvccdo7s0.1sr;received=147.135.0.128
From: "Somewhere TX"
To: "John Doe"
Call-ID: SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002
CSeq: 971341210 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Type: application/sdp
Content-Length: 184
v=0
o=root 2470 2470 IN IP4 192.168.1.224
s=session
c=IN IP4 192.168.1.224
t=0 0
m=audio 19986 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
---
-- Executing Wait("SIP/NNNNNNNNNN-e8c4", "2") in new stack
asterisk1*CLI>
<-- SIP read from 147.135.0.128:5060:
ACK sip:s@192.168.1.224 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK4ael0810dodhrbsfe3s1.1sr
From: "Somewhere TX"
To: "John Doe"
Call-ID: SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002
CSeq: 971341210 ACK
Contact:
Max-Forwards: 69
Content-Length: 0
asterisk1*CLI>
--- (9 headers 0 lines)---
-- Executing Playback("SIP/NNNNNNNNNN-e8c4", "ss-noservice") in new stack
-- Playing 'ss-noservice' (language 'en')
asterisk1*CLI>
<-- SIP read from 147.135.0.128:5060:
BYE sip:s@192.168.1.224 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK4ael0b10covg3e41q601.1sr
From: "Somewhere TX"
To: "John Doe"
Call-ID: SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002
CSeq: 971341211 BYE
Max-Forwards: 69
Content-Length: 0
--- (8 headers 0 lines)---
Sending to 147.135.0.128 : 5060 (NAT)
Transmitting (NAT) to 147.135.0.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK4ael0b10covg3e41q601.1sr;received=147.135.0.128
From: "Somewhere TX"
To: "John Doe"
Call-ID: SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002
CSeq: 971341211 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0
---
Destroying call 'SD28kl001-677199a19e830060a1ec7072cd4599c8-js11002'
asterisk1*CLI>
Hope this helps.....
Re: Settings for Broadvoice ? HELP!!
Sun, 11/19/2006 - 2:15am
OK, I think I am one of the dumbest people around. At one point I thought I was fairly smart, but after working with Trixbox, I decided I am not. Here it goes:
I installed Trixbox on an older PC (1.2, 512mb RAM, 80Gb)
I have 5 extensions setup and working for internal calls.
I have copied and pasted every bit of coding I have come across in these forums. I am to the point right now were I am pretty sure I have bits and pieces from everyone's help-me files.
I would like to get this set up for my house. I have 2 teenagers, and the phone ringing non stop is driving me crazy.
What I (think) would like to do is the following:
When someone (their friends) call a phone number, they will be greeted with a Prompt. For Child 1 press XX for Child 2 press XX, etc..
It will then transfer the call to their Soft phone. If they are not their it will go to voicemail.
If they need to make a phone call they would dial from their PC and that would be all.
I am at the point were I would be willing to pay/trade someone to help me get this going.
For trade I have the following. I run a few websites for my self and friends and family. It was cheaper for me to become a reseller, then hosting for each domain and site. I have allot of space that I am not using, as well as domain purchase credits. If anyone would like to have web space (Must follow my TOS policy) I could trade that.
or.. Good old fashion cash (Please be gentle though. I do have 2 teenagers)
Let me know.
Re: Settings for Broadvoice ? HELP!!
Sun, 11/19/2006 - 2:41pm
smartin74, you should have started a new thread instead of posting in the "settings for broadvoice" thread, that is, unless you need help with that also :) That way you might get more people to help.
Anyway, sounds like you need to set up IVR. In FreePBX, go to IVR (I think it may have been listed as 'Digital Receptionist' in earlier version of FreePBX). Then, you can set up what you are talking about. Actually you'll have to go to system recordings also, where you will record the "Dial X for ..." message, then in the IVR you can select which message to use. You can check the box by "enable direct dial" so people can dial the extension of whoever they are calling right away. If you have just a few extensions you could list them in the message, for example "Dial 200 for Billy.. etc". Or, you can enable the directory where someone would push # to choose the person they want to call.
I hope that was able to help a little bit!
Re: Settings for Broadvoice ? HELP!!
Sun, 11/19/2006 - 7:30pm
Sorry if I posted in the wrong spot. Was searching for the settings when I cam across that post.
Actually I think I may need help with everything. At one point I had it accepting calls, then when I rebooted it (Need to get away from doing that, since it is not Windows) it stopped working. I tried to change a few things and then everything stopped working. I am going to do a fresh install and see what happens. My offer is still out there for anyone that can help me set it up.
Would it be wrong to copy my post and set it in a new thread? Would that be considered flaming?
Thanks
Steve
Re: Settings for Broadvoice ? HELP!!
Sun, 11/19/2006 - 8:19pm
I would go ahead and start another thread in the "Help" section. Things could end up getting too lost/too confusing in this post. Do you have Broadvoice, by the way? I just set up a broadvoice trunk the other day so I could probably help you there.
PS: You probably dont really need to do a fresh install. Trunks, extensions, etc. can be deleted and re-created if needed. If you reinstall you will still be left with the same things to learn. :)
Re: Settings for Broadvoice ? HELP!!
Mon, 11/20/2006 - 10:48am
I am new to Trixbox but I am in process of trying to set up a configuration using Broadvoice for an offsite business location. I must say I have been getting a bit confused but all the various settings that are supposed to work using Broadvoice. At this point I have a couple of problems: 1)When I call in it just rings 2)When I call out I get an "all circuits are busy" msg.
Not sure what info would be needed bu I can proivide any info as requested, thanks in advance.
Re: Settings for Broadvoice ?
Tue, 01/02/2007 - 2:25pm
I could use some help on this one. I have my broadvoice conenctions setup and running. Incoming and outgoing both work fine, but half of the time on incoming calls the IVR does not register the buttons being pushed to get to an extension. Even when I set it up for just a single button press "press 0 to go to the operator" I can hit "0" 10 or 20 time before it accepts it and sometimes it doesn't accept it at all. At the moment I have all the broadvoice calls going directly to the receptionist, but I would like to go to my main IVR. Has anyone experience this? Any suggestions?
Thanks
David Wolff
Mines fixed now
Thu, 05/03/2007 - 10:50am
I found my problem. It was with the /etc/hosts file.
I had sip.broadvocie.com defined as 147.135.12.128 and
I had proxy.nyc.broadvoice.com defined as 147.135.20.128.
It seems that since sip.broadvioce.com is explicitly stated in the .conf files it was never getting to the correct gateway (the one I wanted was NYC) to register.
I changed sip.broadvoice.com to the nyc address, 147.135.20.128 and it registered right away.
I guess I should have RTFM a little closer. Hope this helps some folks who are misdirected by trying to fix their sip.conf files when there is nothing wrong there.
S4SMB
Broadvoice Nightmare Saga- FINALLY Get Incoming Calls
Sun, 05/13/2007 - 8:44am
I FINALLY have Broadvoice setup and working on Trixbox 2.2
I will try to make this as easy to understand as possible, for those who are as green as I am.
Here's the saga...
I had TB 2.0 installed with Broadvoice as a primary SIP trunk and Junction Networks as a backup. All working well. Being overanxious and overenthusiastic, I tried to upgrade to 2.2 too early, made a mess of things, and finally downloaded the 2.2 ISO and started from scratch. (WRONG thing to do, BTW)
I followed all the Broadvoice instructions just as I did with 2.0, only this time, I couldn't get the damned thing to register. I used netconfig to set a static IP and to point to my NAT as the gateway. I edited the /etc/hosts file as instructed, choosing first theNYC proxy and then the DC proxy- no good. Finally, even though my trixbox is behind a NAT, in ignorant desparation, I wiped the sip.nat.conf, leaving it entirely blank, and BOOM, Broadvoice miraculously registered.
I set up the Broadvoice trunk following the BV instructions precisely-but could not receive incoming calls. I tried every single alternative configuration I could find on the internet-no good. Though I could make outbound calls, I could not receive incoming calls- they went to an Asterisk error message. I couldn't see the calls coming in on the CLI but I COULD see the calls hitting my Trixbox when I ran SIP DEBUG from the CLI, getting a message that said "Cseq ACK 1" (whatever that is) and going straight to an error message "The party you are trying to reach is unavailable....".
If I went to the Broadvoice website, logged into my account and forwarded the incoming calls to my Junction Networks DID... THEN the calls came in fine to the Trixbox, and went through to the proper extension using the incoming route I created ("Any DID/Any CID")
I forwarded ports 5004-5080 on my router to individual ports on my Trixbox (instead of the way I had it before, where the entire range of ports were forwarded to port 5060 on the Trixbox), using UDP, of course (MAKE SURE YOUR PORTS ARE FORWARDED USING UDP AND NOT TCP) - and THEN I could see the incoming Broadvoice calls hit the CLI (without running SIP DEBUG) but the calls still got the "Cseq: 1 ACK" treatment and got the same error message.
After HOURS upon HOURS of trying to fix it (with the patient, kind, gentle and noble assistance of several members of this community) the thing that FINALLY made it work was re-appending the DID to the end of the registration string after a forward slash ("/"), thusly:
my_BV_DID@sip.broadvoice.com:my_BV_SIP_password:my_BV_DID@sip.broadvoice.com/my_BV_DID
BOOM!
All working fine! (for now, at least)
For the record, here's how to set up using my settings, in case it helps:
(I assume that you have created a BYOD Broadvoice account and that you have added at least one working extension to your Trixbox)
SIP TRUNK settings from FreePBX
Outbound Caller ID: Your_BV_DID (Your Broadvoice number- just type the phone number in by itelf)
Never Override Caller ID: Leave unchecked.
Maximum Channels: 2
Outgoing Dial Rules: Leave blank
Outbound Dial Prefix: Leave blank
OUTGOING SETTING
Trunk Name: Broadvoice
PEER Details: (see below)
authname=Your_BV_DID (Your Broadvoice phone number-just type type the number straight in after the "=")
canreinvite=no
context=from-broadvoice
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=Your_BV_DID (Your Broadvoice phone number)
host=sip.broadvoice.com
insecure=very
nat=yes
secret=Broadvoice Password (This is NOT the password that you use to log into your Broadvoice account! This is the ten-character password that is found by clicking on the "account settings" link from the "Account" tab AFTER you log into the Broadvoice website using your account password)
type=peer
user=phone
username= Your_BV_DID (Your Broadvoice phone number)
INCOMING SETTING
USER Context: sip.broadvoice.com
USER Details: (see below)
context=from-pstn
dtmf=rfc2833
dtmfmode=rfc2833
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=Broadvoice_SIP_Password (CAREFUL!! This is NOT the password that you use to log into your Broadvoice account! This is the ten-character password that is found by clicking on the "account settings" link from the "Account" tab AFTER you log into the Broadvoice website using your account password)
type=friend
user=Your_BV_DID (Your Broadvoice phone number-
username=Your_BV_DID (Your Broadvoice phone number)
Register String:
Your_BV_DID@sip.broadvoice.com:Your_BV_SIP_PWD:Your_BV_DID@sip.broadvoice.com/Your_BV_DID
Click the "Submit" button at the bottom and THEN DO NOT FORGET TO CLICK ON THE BAR AT THE TOP OF THE PAGE TO RELOAD THE CONFIGURATION.
To set up an Inbound Route (from the FreeBPX "setup" tab):
Leave EVERYTHING blank or in its default state EXCEPT for the "Set Destination" settings. There, pick an extension (If you haven't made an extension, you will need to make one first) and check it from the "Core" menu.
Click the "Submit" button at the bottom and THEN DO NOT FORGET TO CLICK ON THE RED BAR AT THE TOP OF THE PAGE TO RELOAD THE CONFIGURATION.
This will create an Inbound Route named "Any DID/Any CID"
FINALLY- in the "General Settings" link on FreePbX "setup" page.. Early on in the saga, I changed "Allow Anonymous Inbound SIP Calls" to "yes" and that is the wa I still have it. I don't know if this makes a difference or not- I have a conference call scheduled for tommorrow AM and I am afraid to chage ANYTHING now that I have things working, but next weekend, I will change it back and see if it keeps my Broadvoice calls from coming in again.
GOOD LUCK!...
I hope this helps somebody!
IT WORKED for like 2 hours...
Mon, 05/14/2007 - 1:41pm
So i had it working with what Paul said but now it's back to not working. But at least now the incoming works but not the outgoing. I'm not sure what happened but all i get from xlite is :
the person you are calling is not available please try again. Call Failed: Not Found
PLEASE HELP!!! I need to get a working number for tomorrow at least. I don't care about features right now. it was working now i don't know what happened. I can't insert the IP for broadvoice proxy either because it won't let me in HOST.CONF this is what i get:
Warning: copy(/etc/host.conf): failed to open stream: Permission denied in /var/www/html/maint/modules/09_configedit/cls_phpconfig.php on line 315
Write failed!
But i don' tthink that's the problem since it was working without it that anyway.
Stupid question. I used one
Sat, 11/24/2007 - 10:33am
Stupid question. I used one of the configs up above. I can make outgoing calls, but for some reason I cannot receive incoming calls.
Any reason for this? Below is an incoming call. Should be routed to extension 100.
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Wallace Michael' number is '3608741957'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 100 to extension map
-- dialparties.agi: Extension 100 cf is disabled
-- dialparties.agi: Extension 100 do not disturb is disabled
-- dialparties.agi: dbset CALLTRACE/100 to 3608741957
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("Local/100@from-internal-0252,2", "SIP/100||tr") in new stack
-- Called 100
-- Local/100@from-internal-0252,1 is ringing
-- Got SIP response 482 "Loop Detected" back from 192.168.15.10
-- Now forwarding Local/100@from-internal-0252,2 to 'Local/100@from-internal' (thanks to SIP/100-b7812d10)
-- Executing Macro("Local/100@from-internal-e9a1,2", "exten-vm|novm|100") in new stack
-- Executing Macro("Local/100@from-internal-e9a1,2", "user-callerid") in new stack
-- Executing NoOp("Local/100@from-internal-e9a1,2", "user-callerid: Wallace Michael 3608741957") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "AMPUSER=3608741957") in new stack
-- Executing GotoIf("Local/100@from-internal-e9a1,2", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing NoOp("Local/100@from-internal-e9a1,2", "TTL: 51 ARG1: novm") in new stack
-- Executing GotoIf("Local/100@from-internal-e9a1,2", "0?continue") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "__TTL=50") in new stack
-- Executing GotoIf("Local/100@from-internal-e9a1,2", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing NoOp("Local/100@from-internal-e9a1,2", "Using CallerID "Wallace Michael" <3608741957>") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "VMBOX=novm") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "EXTTOCALL=100") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "CFUEXT=") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "CFBEXT=") in new stack
-- Executing Set("Local/100@from-internal-e9a1,2", "RT=") in new stack
-- Executing Macro("Local/100@from-internal-e9a1,2", "record-enable|100|IN") in new stack
-- Executing GotoIf("Local/100@from-internal-e9a1,2", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("Local/100@from-internal-e9a1,2", "recordingcheck|20071124-093031|1195925431.1209") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- Got SIP response 482 "Loop Detected" back from 192.168.15.10
recordingcheck|20071124-093031|1195925431.1209: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("Local/100@from-internal-e9a1,2", "No recording needed") in new stack
-- Executing Macro("Local/100@from-internal-e9a1,2", "dial||tr|100") in new stack
-- Executing GotoIf("Local/100@from-internal-e9a1,2", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing AGI("Local/100@from-internal-e9a1,2", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
I get this with the command " asterisk -vvvr "
I'm still new at this.
Thanks in advance.
~Michael
Thank YOU PAULV
Fri, 08/15/2008 - 10:13pm
I cannot thank you enough! I have spend 20 or 30 hours trying to get my trixbox running with BV! I have done numerous reinstalls (to clear settings that I may have missed on reset), to trying literally hundreds of combinations of options that would work!
As most people have found; they end up using snippets of "options" and ideas deposited by the many, finding that outbound might work, inbound fails, or the other way around.
I followed your instructions to the T, as I did with all the others, but it was yours, and only yours that worked! Thanks a billion!
Many Thanks PaulV, your step
Mon, 04/06/2009 - 3:26pm
Many Thanks PaulV, your step by step process worked, as I had to spend 3 hours + to make the pbx accept incomming calls with vain. NOw it is working.. Thank you PaulV (for the post Sun, 05/13/2007 - 8:44am)
Freed
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Member Since:
2006-06-01