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Callmanager SIP-Trunk to Vonage softphone 1

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tconn

tconn

IS-IT--Management
Nov 30, 2001
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Is there anyone out there who has successfully configured a sip-trunk from a Cisco Callmanager to a Vonage softphone account?

Anyone?
 
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Finally, everything is working and working good.

I can place calls via our IP Phones and recieve incoming calls to our IP Phones. I had to enable "sip debug peer sip.broadvoice.com" to see what was going on.

After some tweaking around the main thing that worked for me was the following:

register => 214xxxyyyy:4qcudfvkjb@sip.broadvoice.com:5060/214xxxyyyy

I had to add the extension (the full e164 number) at the end of the register command because I needed that full number to be sent to my CCM4.1.2. From there my CCM4.1.2 will use only the four last digits and forward the call to a set of shared lines.

So, everything is working. Thanks to you tconn! I really do appreciate your help on this.

So one last quesion or opinion, is this solution, involving Asterisk PBX as a middle piece, working out for you in terms of quality and performance?

Thanks again. You are my hero.

ciscofreak
 
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hey tconn, i need your help again. no changes were made, but now I cannot recieve inbound calls. outbound calling is still good. Here is a copy of my extension.conf file for inbound calling. Does this look right compared to what you have working?

[context]
exten => _NXXNXXXXXX,1,dial(SIP/${EXTEN}@cmp01tra,30)
exten => _1NXXNXXXXXX,2,congestion()
exten => _1NXXNXXXXXX,102,busy()

thanks!

cf
 
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The first line doesn't match the others. Your incoming calls probably are sent to 1NXXNXXXXXX rather than just NXXNXXXXXX.

You might look at making a rule like "anything inbound":
exten => _.,1,1,dial(SIP/${EXTEN}@cmp01tra,30)
exten => _.,2,congestion()
exten => _.,102,busy()


 
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tconn, could you send me your config files, minus the logins of course.. I've been trying to seutp asterisk as a proxy for my callmanager to my broadvoice account but wasn't having much luck..
 
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It would be easier if you sent us your config or even the logs...

<Sip.conf>
register => 2145551212:pw@sip.broadvoice.com:5060
[sipBroadvoice1]
type=peer
user=phone
host=sip.broadvoice.com
fromuser=2145551212
fromdomain=sip.broadvoice.com
secret=pw
username=2145551212
insecure=very
context=main-menu
authname=2145551212
dtmfmode=inband
dtmf=inband
canreinvite=no
nat=yes
qualify=yes

[labcm33]; THIS IS MY CALLMANAGER
type=friend
host=1.2.3.4
port=5061; I use a nonstandard port, not normally needed.
context=labcm33
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=no
qualify=yes

<extension.conf>
[labcm33]
exten=>_NXXNXXXXXX,1,goto(outbound,${EXTEN},1)
exten=>_1NXXNXXXXXX,1,goto(outbound,${EXTEN},1)
exten=>_9NXXNXXXXXX,1,goto(outbound,${EXTEN},1)
exten=>_91NXXNXXXXXX,1,goto(outbound,${EXTEN},1)
exten=>XXXX,1,dial(SIP/${EXTEN}); inhouse call

[outbound]
;check for an available channel at BV, fallback to PRI if Broadvoice isn't taking calls today...
exten => _.,1,noop()
exten => _.,2,ChanIsAvail(SIP/${EXTEN}@sipbroadvoice1)
exten => _.,3,Dial(SIP/${EXTEN}@sipbroadvoice1,,${OUTBOUND_DIAL_OPTIONS})


The settings for the callmanager are usually straightforward. Nothing fancy on that side...

Troubleshooting hints:
1. Can you call your BV number and see activity on your asterisk console? If not, you aren't registered properly.
2. Do you see console activity on outbound calls? Add some 'exten=>...NoOp(This is debugging info. Call to ${EXTEN} in the context: labcm33)' debugging info...

I am the first to admin, I spent many many fruitless hours farting around with different settings until it finally worked. In the end, it was the most simple config that seemed to work best.

 
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Okay, got outbound calls working...

Does broadvoice send me digits that I need to pass along to the CCM so it can route?

Also, I'm assuming there is a 'context' that i need to forward to.. Any help on that?
 
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Pass along? No.. But you do need to dial whatever number you want the call to goto:

[main-menu]; this is the context I used in the sipBroadvoice entry above
exten => <BVnumbr>,1,Dial(SIP/1001@labcm33); will send the call to extension 1001 on the CM.

You might want to look at the pages on They have a a wealth of info on these types of issues.
 
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Sweet.. Had another issue that I got resolved..


Thanks for your help!!!


I have another question for you.. Are you allowed to make multiple calls at the same time? I can do that.. I would figure that broadvoice would deny this feature.. Espsecially since I have the cheap plan right now...
 
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Well, you know ... before I was able to get inbound and outbound working through broadvoice.com, but 24hrs later inbound calling STOPPED! Nothing was changed in terms of my configuration or setup. I tried to speak with broadvoice support ...no response. that is unaccepable to me.

So, I tried using VoicePulse Connect and that has been working out for me the last few weeks. That is how it should work. Plus there tech-support is awesome with timely responses.

So, I saw your last post with your entire configuration (sip.conf, ext.conf) and attempted to use that to re-configure and connect with Broadvoice. I get the same the problem. Outgoing calls are GOOD, Inbound Calls are "BUSY" and doesn't work.

So, honestly I am not sure if you gentlemen are doing something more that isn't listed here, but inbound calling doesn't work. The console shows "BUSY". I switch my config back to VoicePulse ... it works! Please share my magic with me and what you are doing to get this to work for inbound/outboud calling.

If I am missing anything or if there is something I need to do please help me because I have wasted hours with SIP to BROADVOICE with no true results. However spending minutes with SIP to VoicePulse.

Any feedback, HELP, and guidance would be great.

Thank you

ciscofreak
 
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I've got this working flawlessly with BroadBoice but I would like to use my Vonage account. Where did you get the Vonage account information such as password and host name for your vonage account? Thanks
 
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Has any one got this working using a wintel solution rather than using asterisk on a Linux platform.
 
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jrlance, what do you have for your sip.conf and extension.conf files, which is working with BroadVoice?

Thanks!

cf
 
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ciscofreak, here are my config files. They are very simple and don't do any real error checking or passing of digits, but it works.

sip.conf

[ccm]
type=friend
context=incoming
host=192.168.100.101 (IP of CallManager)
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

[bv]
username=########## (Broadvoice number here)
user=phone
type=peer
secret=########## (Broadvoice password here)
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=########## (Broadvoice number here)
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=to-ccm
canreinvite=no
authname=########## (Broadvoice number here)

[sip.broadvoice.com]
username=########## (Broadvoice number here)
user=########## (Broadvoice number here)
type=user
secret=########## (Broadvoice password here)
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=to-ccm

extensions.conf

[incoming]
exten => _NXXNXXXXXX,1,goto(outbound,${EXTEN},1)

[outbound]
exten => _.,1,noop()
exten => _.,2,ChanIsAvail(SIP/${EXTEN}@bv)
exten => _.,3,Dial(SIP/${EXTEN}@bv,,${OUTBOUND_DIAL_OPTIONS})

[to-ccm]
exten => _.,1,Dial(SIP/1001@ccm,30) (rings extension 1001 on my call manager)
 
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I'm useing Asterisk 1.0.8 and CallManager 4.0.2(a) H.323 works fine but I'm having problem setting up SIP trunk.

I'm getting the error from the Asterisk

SIP/2.0 400 Bad Request - 'Malformed/Missing URL'

Does anyone know what the problem is?

Also what version of the CallManager are you guys running which work with the asterisk?

Thanks
Charles
 
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Does anyone else get bad call quality with broadvoice? I get garbled voice, etc.. But I have a vonage account as well and it works fine. I did change to like the LAX proxy and it worked better.. Seems like broadvoice is running out of bandwidth.


thoughts?

BuckWeet
 
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BuckWeet, do you have your Asterisk server connected to Vonage like you do for Broadvoice via a SIP or IAX trunk?

cf
 
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No, i just have a phone adapter for vonage... then using an FXO port on a router..
 
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