Cisco Support Community

Problem with voip call between two sites


I have a little problem with a voip call between two sites, i am new at VoIP.

First let me explain my scenario,

I have two sites, each at different cities, they are connected using a Cisco site-to-site VPN, Site A (Cisco 2811), Site B (Cisco C1861-UC-4FXO-K9).

The vpn is working perfectly.

I could dial from a sip phone (ext: 15 at Site A) to a sip softphone (ext 112 at Site B), (neither sip phones are cisco) without problems, but i could not dial the opposite, when i dial from ext 112 at site B to ext 15 at site A, i got server error on the sip softphone display.

I don't know what is wrong with the conf, this is the dial-peer configured on site B CME to dial to site A.

dial-peer voice 10 voip

destination-pattern 1.

session target (I'm using the LAN IP address)

codec g711ulaw

This is the dial-peer configured on Site A CME to dial to Site B

dial-peer voice 100 voip

destination-pattern 1..

session target (I'm using the LAN IP address)

codec g711ulaw

I could ping both LAN IP address from the peer router.

I have captured the show debug voice dial-peer on both CME (file attached), and i understand that the call is transfered to CME site A, but i don't know what is going on after.. i see something with ext 11, but i dont understand what it means, (ext 11 is the operator, num-exp 0 11).

Hope somebody have any ideas about what could be wrongly configured


Juan Pablo

Everyone's tags (5)
Community Member

Problem with voip call between two sites

If all devices are SIP then you should have this on your dial peers

session protocol sipv2

dtmf-relay rtp-nte

Any reason you arent just running long locals off your CUCM cluster and using SRST at the remote site. Using CME for a small number of phones increases your complexity.SRST will support SIP to PSTN and you can control the 1861 via MGCP with the CUCM until the need for fallback.

CreatePlease to create content