I have a SIP line that is working fine when I make calls from IP
phones. I can send and receive calls. The problem is that if I try to dial from the CLI using the originate command or use an AMI connection to originate a call I get the following error:
originate SIP/protel-out/0445540881644 application playback tt-monkeys WARNING: chan_sip.c:20437 handle_response_invite: Received response: "Forbidden" from '"Anonymous"
Here is the sip.conf entry for that line:
[protel-out] defaultuser=XXXXXXXXXX secret=XXXXXXXX fromuser=XXXXXXXXXX type=peer fromdomain=i2next.com.mx host=i2next.com.mx disallowed_methods = UPDATE nat=no qualify=no insecure=port,invite directmedia=no disallow=all allow=g729 context=entrada trustrpid=yes sendrpid=yes
As I mentioned it works if I dial from…
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…
I am using Red Hat Enterprise Linux 5.5 (32bit) and Asterisk 1.8.12, Dahdi 126.96.36.199 and libpri 1.4.2. The installations is fine. But in the Asterisk CLI prompt the pri commands are missing, only the pri intense debug span is populated. Even if i execute that command it results me to "pri set debug 2 span 1 is not a valid command"
Any assistance would be appreciated.