I have a vonage dial tone that connect to a fxo port on a cisco(2821). The cisco router connect to a remote 2621XM and a NEC PBX. Ihave programed a dial dial that allow a remote user to dial a code for Ex123 to pull a dial tone from the Vonage adapter and dial out. The problem I am facing is Capacity. Since I can only get 4 FXO per card I need a large amount of FXO ports 50 in order to provide this service to remote offices worldwide in my company. I tried to experiment with an FXS ports since I can order the 12 ports module and I found the minute I plug the Vonage Analog port to the FXS ports it goes offhook and if you issue the show voice call, it shows in park mode. Is ther a solution where I can use the FXS ports instaed the FXO ports. Any Configuration hints will be great. Any Technology Advise on this type of setup will be gratly appreciated.
You cannot plug a trunk line such as vonage into an FXS port, no tricks, it just won't work.
If you really need 50 lines I would not recommend Vonage. I would look at using a few PRI digital circuits (46 lines) as it would only require a single 2 port T1 voice card.
You could also investigate providers of SIP dialtone. I know that Level 3 provides dialtone for companies via SIP and is geared for the type of solution you seem to be looking for. This would be similar to Vonage as it would be IP based dialtone.
You had posted this question already two days ago, here again my reply that goes inline with the other answer you received.
this is the business plus service:
it also avaialbe from other resellers, just search. It uses SIP so you would just place one router interface on the internet, and place calls. You may need an IP-to-IP images on the 2811 to do this, on the other hand, if the 2600XM has an internet connection, it could place the calls directly as well.
However, for testing also a "vonage softphone account" should work, altough I think it wold not let you place multiple calls.
As I said, it would be very expensive to equip a cisco routers with that many FXO ports, contact your reseller and verify yourself.
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Thanks for your reply,
I am using FXO ports now. You mentioned I Place one router interface on the internet, How this will work since I am using voip dial-peer on the remote routers to pull a tial tone from the vonage FXO in a hub and spoke configuration. Do I need any additonal hardware other than the router that connectto the internet. Thank you for any additional explanation since I am confused at this.
once your are set with a SIP provider, just configure a dial-per for it:
dial-peer voice 20 voip
session protocol sipv2
session target ipv4:184.108.40.206
authentication username xxx password zzz
You will need an IP-to-IP GW image if you want to relay calls from other voice GWs. This is called a BSC.
Else, you can give internet connectivity to the other voice GW as well and they will directly place calls to the SIP provider.
It is really much simpler and economic than setting up a bunch of FXOs and single-line ATAs.
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In my case I have a seperate Internet service provider for the vonage service so it will not interfere with the Data side. I have the remote users that has a legacy Nec PBX with dial peer to connect to the HQ. On the remote router I have defined a dial peer to point to the Vonage router on the HQ. How do I have an exit from the vonage router to the internet if I dont have pots configured?. or in this case I am not using pots config!!!!. I am trying to undestand it better. Thanks again for the help
Hi, as I said before, the call will be made with IP / SIP so once you configure the dial-peer it will start working. Yo do not need pots for this to work.
Also as I said before you either need an IP-to-TP image (these which filename ends with _isv), or configure the router connected to the PBX with said dial-peer abovea it will connect directly to to TISP, vonage or other.
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Thanks for your help. From your experience do you know of anyone that has this type of setup that did not run into major issue in configuration?. Also will you be kind engouh to mention other provider that seems more stable. If you cannot I understand. Thanks for your help. I will give it a try the way you have mentioned above.
yes many people is using SIP to ITSP without issues. The provider you choose is better to be a reputable one. I think Vonage, Broadvoice, Voicetrading.com are all reputable ITSP. cisco is supposed to be working on a list of "certified" providers to be published.
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Could i have more than one ITSP connected to one router? If yes i need to config?
Could i have CCM 5.1 conected directly to an ITSP?
Yes you can have the router connect to more than one ITSP, however currently there are serious limitations with digest authentication, first you are limited to a maximum of 5 sets of username/password/realm sets, second all of them must have the same username!!!
I'm not expert with CCM 5.1 but I'm sure someone else will be able to help.
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Thanks for quick answer.
But in router under sip-ua i can only
enter one time authentication username XXX password xxx am i correct?
How can i config the others?
As I mentioned before, you can have up to five of these sets, username must be always the same, password and realm can be different. So when connecting to multiple ITSP, either you manage to be assigned you the same username, or they must not use digest authentication at all.
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thanks for help.Is the Dial-peer below is taking from a working config?. Is this all I need on the Vonage router with the _isv image to start making assuming I have regitered with an ISTP?. Do you know of a case on Cisco site or on the web that has done this type of setup or cloase to it to look at before I start going this direction?. Thanks again for your help you have been very patient man.
One more thing I have found the below on sipdiscount for testing. you have mebtioned in the dial-peer I have to have the IP address of the far side of the SIP server. In the below case in order to find it just ping the server name and by trial and error find the correct IP address as you have mentioned above.
User Name: test
SIP Proxy/registrar: sip.sipdiscount.com
SIP Outbound Proxy (optional): sip.sipdiscount.com
STUN server (optional): stun.sipdiscount.com
I don't know if you are using sipdiscount.com already, but in the recent past they were blocking cisco routers at least for trial users. Don't know if that has changed now.
The configurations on the router depends on various things and may need modifications. with the IP-to-IP GW images, for the dial-peer that handles the remote GW, you will need "codec transparent". On the other hand, it will need "codec g729" for calls coming from local ports.
Anyway, once you have things more or less running, come back here in case of problems.
Under sip-ua i tried to enter more than one
authentication username xxxx password xxxx without sucess. What minum IOS required?
And under sip-ua i can also config one sip server and one registrar don't i need to config one for each ITSP?