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New Member

SIP Trunks to Broadvoice

Is it possible to have multiple "trunks" to a provider like Broadvoice? I am asking about outgoing calls specifically. Since a username and password are required, where is that setup?

I am asking for both CallManager (4.1) and IOS. In IOS, i can't see where you can configure more than one login under sip-ua

Anyone done this before?

Steve

4 REPLIES
New Member

Re: SIP Trunks to Broadvoice

Anyone?

Blue

Re: SIP Trunks to Broadvoice

I got this from another forum where the poster had gotten it working along with MWI:

--------------------------------------

I've already recieved an email asking for the config, so I guess that officially makes it interesting:

Here are the relavent parts of the config I used to get CCME working with BroadVoice:

Obviously the items in the config that begin with YOUR should be replaced with the IP addresses/names of the appropriate items for your setup.

I used a sample BV number: 1234567890 and password YOURSIPPASSWORD in the config below.

The sticky points were:

- Using sip.broadvoice.com for my dial peer targets, not proxy.broadvoice.com

- Using "dtmf-relay rtp-nte" under the dial peer to get DTMF working

- Using "no remote-party-id" under sip-ua*

Getting MWI (Message Waiting Indicator) to work involved:

- Adding the "mwi-server" with the "unsolicited" keyword to my sip-ua

- Adding "mwi sip" to the ephone-dn's that you want the MWI light to work

Things to note:

- I used the "voice class codec" in my dial-peers to allow the use of multiple codecs

- If you just want G.711U, then just add "codec g711u" to your dial-peer and skip all the "voice class codec" stuff

*Later versions of IOS add the "Remote Party ID" SIP header. They put phonenum@gatewayipaddress in the field. BV (and any SIP compliant provider) replaces the From information with info from this field. Well, BV doesn't like From fields to end in anything but sip.broadvoice.com (well, maybe a couple of others, but certainly not the IP address of you router!). So by inserting this command, it suppresses the creation of this field and the BV proxy will use the From field. This item is what gave me the biggest headache. IOS debug commands are your friend

By the way, the "sip-ua" and dial-peer parts of this config should work for any generic IOS device (later than 12.3(T or so).

Lastly, there is some extra fluff in the dial peers. I turned off VAD and set IPP to 5 for my RTP packets. These are not necessary but I've found them to be useful additions.

I hope this helps someone!

-Nick

 

ip domain name YOUR.DOMAIN.NAME

ip name-server YOUR.DNS.SERVER

!

!

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

!

ip route 0.0.0.0 0.0.0.0 YOUR.DEFAULT.ROUTE

!

!

!

dial-peer voice 1 voip

destination-pattern 1..........

voice-class codec 1

session protocol sipv2

session target dns:sip.broadvoice.com

dtmf-relay rtp-nte

ip qos dscp cs5 media

no vad

!

!

dial-peer voice 86 voip

destination-pattern *86

voice-class codec 1

session protocol sipv2

session target dns:sip.broadvoice.com

dtmf-relay rtp-nte

ip qos dscp cs5 media

no vad

!

sip-ua

authentication username 1234567890 password YOURSIPPASSWORD

no remote-party-id

mwi-server dns:sip.broadvoice.com expires 3600 port 5060 transport udp unsolicited

registrar dns:sip.broadvoice.com expires 3600

!

!

telephony-service

voicemail *86

!

!

!

ephone-dn 1 dual-line

number 1234567890

mwi sip

!

!

!

!

ephone 1

mac-address XXXX.YYYY.ZZZZ

button 1:1

!

!

ephone 2

mac-address XXXX.YYYY.ZZZZ

button 1:1

!

-----------------------------

please rate helpful posts.

adignan - berbee

New Member

Re: SIP Trunks to Broadvoice

i saw this post as well. this only has one account with a provider. i was looking for a config with multiple accounts.

Thanks

New Member

Re: SIP Trunks to Broadvoice

Hi Adignan,

I'm trying to setup a sip trunk to other sip provider, using a sip softphone client its working my account,

When i config CCME (using SCCP phones) i have not success:

-i see in show sip-ua registar status that line is registered but i when i try to call a number that they(sip provider) provide to test connectivity i hear your account is not authorized, so i'm reaching sip provider but something is not ok, could remote-party-id command help? (i will test!)

In debugs i only see this header (remote-id-party=mymumber@gwipaddress) in INVITE message and when i receive message 100 Trying and message 183 Session Progress from header its seems ok mynumber@sipprovider.com

I am also behind a firewall with NAT.

One more question in sip-ua we don't need also sip-server command?

Tanks

Best Regards,

MC

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