08-01-2012 07:14 AM - edited 03-16-2019 12:29 PM
Hi,
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 ipv4:xxx.xxx.xxx.xxx (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 ipv4:xxx.xxx.xxx.xxx (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
regards,
Juan Pablo
08-06-2012 05:31 AM
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.
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