* You are viewing the archive for August 2nd, 2012

Asterisk Realtime Don’t Support ‘n’ As Extension’s Next Priority

Hi Team,

I want to used *’n*’ as priority in asterisk realtime but asterisk don’t support n as next priority….

I am using Asterisk 1.4.41

Originate Call From Cli Does Not Work For SIP Line…

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[12950]: chan_sip.c:20437 handle_response_invite: Received response: “Forbidden” from ‘”Anonymous”
;tag=as79fffc8d’

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 a phone. CLI or AMI fails.

DTMF Transmission Problem

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:

SIP –
[general]
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
defaultexpiry=3600

[authentication]
[test-voice]
type=friend host=dynamic secret=not_my_secret context=users disallow=all allow=ulaw nat=yes directmedia=no qualify=yes trunk=no

IAX2 –
[general]
bindport=bind_here bindaddr=to_this_address delayreject=yes disallow=all allow=g722
allow=ulaw allow=gsm jitterbuffer=no encryption=yes

[test-fax1]
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.

Thank you,

Noah Engelberth System Administration MetaLINK Technologies

Can’t Get Libpri/PRI To Work

Hi,

I am using Red Hat Enterprise Linux 5.5 (32bit) and Asterisk 1.8.12, Dahdi
2.4.1.2 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.

Regards, Gopal.