Callmanager SIP-Trunk to Vonage softphone 1

[tconn]

Is there anyone out there who has successfully configured a sip-trunk from a Cisco Callmanager to a Vonage softphone account?

Anyone?

 

[BuckWeet]

I would like to know if this works myself


BuckWeet

 

[tconn]

I will probably end up using a asterisk server in the middle. I'm not having much luck doing any trunking from the CM...

 

[tconn]

I'm able to dialout, just not having much luck dialing in... I suspect my firewall is not allowing rtp through.

 

[GatekeeperDan]

Can we keep this thread going. With all the new broadband voip providers, CallManager at home is a really cool option. I am currently using an FXO card in a 1760 for PSTN access. I really want a robust clean solution that allows multiple inbound/outbound calling. A SIP trunk to a provider like Vonage or Packet 8 for a monthly charge sounds like an awesome option. Can anyone post further info on configurations for some of the providers you've tried and had success?

 

[ciscofreak1241]

tconn,

what configuration and setup did you do on CallManager to be able to dial out through the SIP trunk to Vonage?

I'm assuming vonage will provide the IP Address, but is there anything they provide that needs to be configured with CallManager like username, passwords, etc?

thanks

ciscofreak

 

[tc0nn]

I ended up using a linux server running Asterisk to route my calls. The callmanager has a sip trunk to the Asterisk server, then the Asterisk server routes call through Vonage.

 

[youcandoit]

i plan to do this too. i have a 2801 ISR with vic2-fxo and 4 port POE module on it's way..I dont however like the idea of having another server in the middle to manage. and i dont know anything about linux.

 

[ciscofreak1241]

Great, thanks! Is configuring Asterisk a challenge or pretty straight forward? Do you know if the software is free or you have to purchase it? If you know of the best URL or resource for that ... it would be great.

Does that include inbound calling too? Or is that still an open issue?

So, you configured a SIP trunk to the Asterisk. Ok, I am looking into configurin a SIP trunk to one of my internal routers and having that route the call to Vonage. The CSPS by Cisco could be another option, but that could be costly.

I will explore all of these options and we can all come together to recommended that "these" are possble solutions for SIP trunking.


Thanks!

ciscofreak

 

[tconn]

Mine does calls in both directions.

Vonage's marketing department makes medium to large businesses shy away. They way they have things priced doesn't scale unless you use less than 500 minutes a month. Check out Broadvoice's plans (unlimited for $30 I think?).

Like any other decently support linux software, there are many GUIs you can install to make life easy. I'm a nuts and bolts kinda guy.. I hate the PIX PDM and still use CLI etc... So I still do most configs from an emacs window while watching the console in another xwindow. The easy way out is to use the gui. If you want a demo, check out Asterisk@Home. Its a CD base live version. No linux experience is really needed.

We're getting a little off topic here, so <snip>...

 

[ciscofreak1241]

That's ok, I am all CLI myself, which is why I have been struggling with CallManager at the very beginning.

Anyhow, I have looked at some things for Asterisk and it comes down to two main files:

sip.conf
extention.conf

I have only a few questions to start with.

On the following sip.conf file:

[callman01]
type=friend
context=incoming
host=10.0.0.1
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

Is the the option "host" the actually IP Address of the callmanager server? And does your sip.conf file look similiar to that for the most part. This is the example with CCM that it gave me. Any help on that would be great.

I am still dissecting the extention.conf before I start asking questions on that.

thanks in advanced

cf

 

[tconn]

Yup, host= name or IP:
[labcm33]
type=friend
host=172.16.0.1
context=labcm33 ;<- calls in from CM goto this context in extensions.conf
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=no
qualify=yes

 

[ciscofreak1241]

Is there more that it needed in the sip.conf file besides the configuration to the CCM? Does there need to be another set in regards to the SIP provider with something like ...

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=443430YYYY
secret=XXXXXXXX
username=4434300465
insecure=very
context=from-broadvoice
authname=443430YYYY
dtmfmode=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
;nat=yes

I not sure if your sip.conf had something like this and your setup is working, so your experience on this is more important. Let me know about that if your config contains more than just that. Or just that.

In terms of the context, in your case labcm33, is just an identifier that is used in the extensions.conf? From the configs I have been looking at on the net, is there any the resemble something close to what you have running?

The dialplan is what I am looking to and which might get me, but I see an example of one I can get an idea how that will work with my enviornment which has 4-digit internal exts.

Thanks a lot for help on this!

ciscofreak

 

[tc0nn]

Here's my BV stuff:

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

[sipBroadvoice1]
type=peer
user=phone
host=sip.broadvoice.com
fromuser=214xxxyyyy
fromdomain=sip.broadvoice.com
secret=4qchvbkjsadh
username=214xxxyyyy
insecure=very
context=main-menu
authname=214xxxyyyy
dtmfmode=inband
dtmf=inband
canreinvite=no
nat=yes
qualify=yes

 

[ciscofreak1241]

cool, thanks. what command can I use to confirm if the Asterisk server is actually registering with the SIP Proxy (in this case BroadVoice)?

cf

 

[tc0nn]

You'll see failed registrations in the log and on the console if it didn't register, or you can do a "sip show peers"

 

[ciscofreak1241]

I am having issues trying to get the Astrix server to Register with the SIP Proxy (Broadvoice)

When I do a "sip show peers" at the CLI prompt it tells me that the status is UNMONITORED and if I do a "sip show registry" there is nothing listed. Maybe that's normal, but I doubt it. Now the sip.conf is responsible for registering with the SIP Proxy (along with CallManager and SIP phones) and the extension.conf is responsible for the the dialplan for outgoing and incoming calls right?
Here is the config that I have for the sip.conf


***** start of file *******

[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context=context
pedantic=no
register => 209XXXYYYY@sip.broadvoice.comASSWORD:209XXXYYYY@sip.broadvoice.com


[sip.broadvoice.com]
type = peer
host = sip.broadvoice.com
secret = PASSWORD
user=phone
fromuser = 209XXXYYYY
username= 209XXXYYYY
authname= 209XXXYYYY
fromdomain = sip.broadvoice.com
context = context
insecure=very
canreinvite = no
dtmfmode = inband
dtmf=inband


[ccm]
type=friend
context=context
host=192.168.1.69
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

***** end of file *******


My extension.conf file has the context [context] for calling, but I am trying to troubleshoot the sip.conf and the registration issue.

Any help would be great?

Thanks

cf

 

[ciscofreak1241]

Ok, I do see the connection now to broadvoice when I do a "sip show peers", should I see something or not listed under "sip show registry"?

Thanks

cf

 

[tconn]

Try mimicking my config on the register line.

 

[ciscofreak1241]

I have been hacking around with this for so long, that I am not slowing down and actually looking at the solution.

Here is the situation and the last piece I will need your help:

- My connection to sip.broadvoice.com has been registered.
- "sip show peers" the status for both shows OK
- I can now call from my IP Phones through the SIP network and communicate with an outside number good.

However, I am having issues with the inbound calling coming in, hence, the extension.conf file. What do I need in my extension.conf file for inbound calling? I remember seeing an example on voip-info.org, but the site is down. Bad timing when I am so close in finishing this. Please let me know what I am missing.

Thanks!

cf