SIP trunking question

tha_dub

2 N00B questions. I just got a 1760....

I have cme 4.1 installed. No pvdm or VIC cards.
Will this support sip trunking to a provider?

Does anybody know of a dirt cheap provider (ideally providing Canadian numbers) that I could use to get this going?

Thanks!

peanutnoggin

I don't see why it wouldn't. I'm not at home to test... but as long as you add your User Agent/SIP user/password Config info... you should be alright. You may need an additional ethernet/fast ethernet port to connect back to your LAN... assuming you're using the built-in Fastethernet to connect to the internet. HTH.

-Peanut

peanutnoggin

Here's an example config that may be able to get you started:

router(config)# sip-ua
router(config-sip-ua)# authentication username <your_sip_user> password <your_sip_pass>
router(config-sip-ua)# sip-server ipv4:<IP_OF_YOUR_SIP_SERVER>

After that... you should only have to configure your dial-peer to route your calls to your SIP server similar to this:

router(config)#dial-peer voice 1 voip
router(config-dial-peer)# destination-pattern 9.......
router(config-dial-peer)# session protocol sipv2
router(config-dial-peer)# session target sip-server
router(config-dial-peer)# dtmf-relay rtp-nte

Hopefully this can help get you started...

-Peanut

tha_dub

Cool thanks!

Now if only I can convince my wife to switch the phone service over completely.... On the other hand I plan to be messing around with this box so much we may only have a phone for a few hours a day

peanutnoggin

tha_dub wrote: »

Cool thanks!

Now if only I can convince my wife to switch the phone service over completely.... On the other hand I plan to be messing around with this box so much we may only have a phone for a few hours a day

That's the way to learn... Let me know how this works out for you... I've only tested this in closed lab environments... but it seems pretty straight forward.

Oh yeah... you'll probably have to throw in:

voice service voip
allow-connections sip to sip
allow-connections sip to h323
allow-connections h323 to h323
allow-connections h323 to sip

And choose your codec as well. I'm still learning this stuff... so I'm prone to errors. Someone with more experience may be able to chime in... Let me know how your configurations go.

-Peanut

tha_dub

While I'm at it. Will these same setting work for a 3cx softphone I'm running on my PC?

Thanks

peanutnoggin

Your softphone should register directly with your SIP service provider. If you want your SIP phone to register with your CME... it introduces a few more configuration steps on the CME. Do you want your CME to provision your softphone or the SIP provider?

tha_dub

I want to provision it to the CME. Basically I'd like to set it up like it would be in a small office where the CME controls everything especially trunk access.

tazdevil

Below is my config;

voice service voip
no notify redirect ip2ip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
header-passing
registrar server expires max 800 min 800
no update-callerid
no call service stop

dial-peer voice 10 voip
destination-pattern 1..........
session protocol sipv2
session target dns:toronto.voip.ms
dtmf-relay rtp-nte sip-notify
ip qos dscp cs5 media
no vad

sip-ua
credentials username xxx password xxx realm toronto.voip.ms
authentication username xxx password xxx
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:toronto.voip.ms:5060 expires 800
sip-server dns:toronto.voip.ms
host-registrar

I can get out, but not in, still working on that part, but out works great.

VancouverTechie

I am thinking of doing the same thing at home. I have a 1751-V with PDVM with FXO, FXS ports.

Anyone in the US recommend a decent SIP provider? I have googled... but nothing really jumps out as a good provider or a good resource for one.

John

peanutnoggin

VancouverTechie wrote: »

I am thinking of doing the same thing at home. I have a 1751-V with PDVM with FXO, FXS ports.

Anyone in the US recommend a decent SIP provider? I have googled... but nothing really jumps out as a good provider or a good resource for one.

John

I've heard some folks mention IPKall but I've personally never used them. If you're looking to just practice for a home lab, I would recommend you build it out yourself using Trixbox. What are your intentions?

-Peanut

networker050184

VancouverTechie wrote: »

I am thinking of doing the same thing at home. I have a 1751-V with PDVM with FXO, FXS ports.

Anyone in the US recommend a decent SIP provider? I have googled... but nothing really jumps out as a good provider or a good resource for one.

John

I think ColbyG has one at home. You might want to PM him to see if he has a suggestion.

pitviper

Pay service or free? For pay, check out Vitelity and Teliax.

VancouverTechie

I am fine with a pay service. I run a small business and would not mind paying for a couple of DID.

Thank you for the info and I will take a look at them both.

John

Anyone have some sample CME, SIP configs?

ColbyG

I use Flowroute, no complaints and it's very cheap. I have a setup guide on my blog if you end up moving forward.

VancouverTechie

ColbyG wrote: »

I use Flowroute, no complaints and it's very cheap. I have a setup guide on my blog if you end up moving forward.

ColbyG,

Thanks, I went ahead with Flowroute and when I get back my office I will configure my router. I also purchased a couple of DIDs and can't wait to test it.

John

VancouverTechie

It is interesting I am able to call out with no issues, but calls coming in is another story. The caller gets a fast busy, and there is no notification on the Cisco 7940 for the incoming call.

I will run some other tests tonight when I get back, but it is progress.