FXS port / SIP Trunk

Can someone please direct me to a sample config of a Pots/FXS port that works with a SIP/ITSP trunk.

most of the sample configs i have found are directly relating to SIP enabled phones and not your standard pots/fxs port.

I also read somewhere that you have to disassociate sccp with the fxs port??

anyhow, much appreciated if someone can direct me to the right document.

fyi.. im testing out my free sip account (unlimited inbound)

so far, everything is registered, but once i make a call, i get an invalid host during debug. This is my first attempt so i know i am missing some configs.

thanks!

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Comments

shodown

your looking at a different technology. When we implement SIP trunks they go right out on a Ethernet port. The FSX ports are analog. SIP is a digital technology

tokhss

Ok, let me rephrase.. I currently have a free ITSP provider thats riding over my broadband connection, shouldn't i be able to use an analog phone with my ITSP ? the ITSP is providing me with a # and i have it assigned to my voice port.

example: ITSP ---WWW---2811---analog port assigned with the ITSP #.

Thanks!

shodown

you could point a dial peer to route the call to the FXS card. You would also have to setup to allow SIP to H323 or whatever you have setup under voice service voip. You may also need media resources if you plan on having any supplementary services going across that trunk depending on the SIP provider

tokhss

Agreed.

the error message i get is "invalid host" during debug messages. everything is registered, sip to sip, sip to h323, h323 to sip, g.711 and g.729 set in the dial peers.

I am not at home. but off the top of my head..

voice-port 0/0/1
signal loopStart
station-id name "sip #"

voice class codec 1
codec pref 1 g711ulaw
codec pref 2 g729

DP Voice 1 voip
voice class codec 1
voice class sip dtmf-relay force rtp-nte
session proto sipv2
session tar dns: "my sip provider"
incoming called-num .
dtmf-relay rtp-nte

DP voice 2 pots
destination-patt "sip #"
port 0/0/1

Thanks! Either way.. if someone can provide me with some example documentation, I would greatly appreciate it. I would really like to cross ref my config against a standard config of some sort.

shodown

SIP is not the best in supported configs, your best best would be to spend some time learning the technology and trying to get it to work the way its suppose to. I work SIP case pretty much everyday and depending on the vendor the configuration has been different. That below is a pretty good start

Cisco Unified Border Element (CUBE) / SIP Trunking Solutions [Cisco Interoperability Portal] - Cisco Systems

UC500.com | Cisco Communication Manager Express, UC520, UC540, SMB VOIP reference and community has bunch of people who deal with these daily SIP battles. I have pretty much thrown my hands up with SIP. I bill project hours if the customer switches to it. Mods delete if its not approved.

tokhss

"By the way its supposed to" ... please explain

.. do you mean .. sip phones type A/B, sip profiles, adding users n all that good stuff..

are analog devices normally neglected with sip setups? lol.

Thanks for the info. Im really just testing it out and trying to learn how to setup a sip ITSP # to an analog device.

shodown

tokhss wrote: »

"By the way its supposed to" ... please explain .. do you mean .. sip phones type A/B, sip profiles, adding users n all that good stuff..

are analog devices normally neglected with sip setups? lol.

Thanks for the info. Im really just testing it out and trying to learn how to setup a sip ITSP # to an analog device.

What I mean by how a technology is suppose to work there are RFC standards that a lot of vendors follow so equipment works with other vendors equipment. The problem with SIP is that when it was new a lot of features weren't supported so vendors added there own flavor of SIP. So even though its SIP its proprietary sip in a lot of cases.

tokhss

Ahhh.. Gotcha.. from everything i read, SIP was suppose to have a lot of momentum and support. It was suppose to dominate and take over H323... from your perspective, has SIP done that yet?

chmorin

tokhss wrote: »

Ahhh.. Gotcha.. from everything i read, SIP was suppose to have a lot of momentum and support. It was suppose to dominate and take over H323... from your perspective, has SIP done that yet?

Well you didn't ask for my perspective, but supposedly SIP was set to be an industry primary many years ago when it was introduced. I'm pretty sure this never happened, and as he said, alot of different 'flavors' were created.

shodown

tokhss wrote: »

Ahhh.. Gotcha.. from everything i read, SIP was suppose to have a lot of momentum and support. It was suppose to dominate and take over H323... from your perspective, has SIP done that yet?

IT should be give it 5 to 10 years and everything will be SIP. I will be posting a long blog on SIP on a case that I have been working for 2 weeks. SIP is great with the amount of calls that you can get over having a PRI. With that said we have companies that are sailing SIP over the internet and you have companies trying to run FAXes modems, DTMF all these other supplementary services without having the people in place to make it work.

tokhss

Chmorin, when i want your perspective, I will ask for it !

LOL.. kidding..

Shodown. thanks for the input. Should have this up and running over the weekend.

pitviper

I'll take a PRI setup over a SiP trunk anyday!!!

tokhss

Pit... lol.. im just trying to learn something new.. i love pri lines.. easy! but something i have yet to do is setup a live sip account .. its free, avail, so why not

tokhss

Following up on this thread..

problem solved. Issue isolated to a NAT config.

what i didnt state in my post was that my sip router was behind edge router.

sip router ip 192.168.254.252 provided by edge (192.168.254.254)

my ACL for troubleshooting sakes was open; host 192.168.254.252 any

during ccsip debug, i noticed the obvious INVALID HOST response, but i also noticed a .241 ip addy .. i checked my edge config, i had an old NAT statement with .241 .. removed that ... still no luck.. so i created a static route to the wan interface on the edge and that worked.

ip nat inside source static 192.168.254.252 interface FastEthernet4

apparently, the sip router needs to be exposed as much as possible.

now its time to buy a block and spread it over multiple sites (routers) !!

tokhss

Here is a basic config thats verified working. I have a free DID account that allows for unlimited inbound calls. I have yet to do any outbound testing.. my ITSP sends 10 digits.

Tools: My cell phone (to call in bound)

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service media-renegotiate
fax protocol cisco
sip
bind control source-interface FastEthernet0/1
bind media source-interface FastEthernet0/1

voice-port 0/0/1
signal loopStart
station-id name
station-id number "ITSP # all 10 digits"
caller-id enable

dial-peer voice 1 pots
incoming called-number .
direct-inward-dial

sip-ua
authentication username ********* password ******* realm asterisk
registrar dns:sipconnect.ipcomms.net expires 3600
sip-server dns:sipconnect.ipcomms.net

hope this helps someone..