I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren’t recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset).
My system set up as follows:
PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE
Asterisk is running Asterisk 10.4.0 on a CentOS 6.2 VM residing on a CentOS 6.3 KVM host. Asterisk has one network interface connected to the Metaswitch without NAT to place/receive calls from the PSTN, and a separate interface to connect to CPE equipment. SIP and IAX are bound to both interfaces. Vocal call quality is fine, DTMF is fine from the customer to the PSTN, but DTMF from the PSTN to the customer isn’t. Asterisk is set to remain in the media path on all calls. The customer facing IP address on the Asterisk server is private and is being 1:1 NATed through a MikroTik RB 1100 to a public address that the customers are then connecting to. I have also placed test calls with the “customer equipment” inside the same LAN as the Asterisk server’s customer facing IP address (no NAT) with precisely the same symptoms. The same symptoms persist whether the PSTN or the CPE initiate the call.
My example configs are as follows:
limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes disallow=all allow=g722
allow=ulaw allow=gsm allowoverlap=no callevents=yes allowguest=no directmedia=no
bindport=bind_here bindaddr=to_this_address srvlookup=yes maxexpiry=7200
type=friend host=dynamic secret=not_my_secret context=users disallow=all allow=ulaw nat=yes directmedia=no qualify=yes trunk=no
bindport=bind_here bindaddr=to_this_address delayreject=yes disallow=all allow=g722
allow=ulaw allow=gsm jitterbuffer=no encryption=yes
type=friend host=dynamic username=test-fax1
secret=not_my_secret context=users disallow=all allow=ulaw qualify=yes trunk=no requirecalltoken=no
SIP peers are Zhone ZNID-2xxx series ONTs. IAX peers are ATCOM AG198 ATA gateways, either behind the ONTs (but on the same voice VLAN the ONTs use to talk to Asterisk) or on my Asterisk server’s local network. The voice VLAN is a different subnet than Asterisk is on, but no NAT exists between the subnets.
Noah Engelberth System Administration MetaLINK Technologies