05-20-2009 06:18 PM - edited 02-21-2020 03:28 AM
Our setup is this. Call comes in via SIP Trunk, routes to CallManager through ASA and back out through to the remote site via VPN. Yeah, it's a complex setup I know.
The problem is that when a caller calls and gets connected with a user, the calling party can hear what the IP phone user is saying but the IP phone user cannot hear anything. If you look at the phone for sending/receiving packets, it is sending but receiving packets are not incrementing. This will happen intermittently (mostly about 75% of the time). The caller will get connected, everything will be working and then the other side goes silent.
My policy map is allowing sip across and the ACL is allowing everything across from the voice subnet. The phones never lose connectivity. I can ping fine from the voice subnet to the remote phone side.
policy-map global_policy
class inspection_default
inspect dns migrated_dns_map_1
inspect ftp
inspect h323 h225
inspect h323 ras
inspect rsh
inspect rtsp
inspect esmtp
inspect sqlnet
inspect skinny
inspect sunrpc
inspect xdmcp
inspect sip
inspect netbios
inspect tftp
inspect icmp
inspect icmp error
Both ASA's are running 8.0(4)29. Any suggestions?
05-22-2009 09:25 AM
Were call come from?
Another IP phone registered on another callmanager?
You must go to de basis of the problem capturing ip traffic in both endpoints. Check source/destination address and check all the path.
Post all information that you can and I will try to help you. I have the same problem a year ago and was a BGP policy routing problem.
Guido.
Please rate all the helpful comments.
05-22-2009 02:10 PM
I would make sure that there isn't any other NAT or ACL's in line of the receive traffic. Then I would take two old pc's and install Wireshark. Using TSHark, capture the SIP, RTP and ICMP traffic into a handful of ring buffer files. When the problem occurs, see if the audio streams are present on the the other side of the ASA. If not, you will want to keep moving toward the other end.
Additionally, you might want to see if your ASA is logging any dropped packets from the other end. In any case Wireshark is a great tool for working with SIP. You can actually listen in the RTP Streams they are using G711.
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