Re: [asterisk-users] Monitoring connection to VoIP provider?

2011-07-12 Thread Steven Stromer
A quick to implement open source network monitoring tool is smokeping: http://oss.oetiker.ch/smokeping/index.en.html Tobi Oetiker and Niko Tyni's awesome tool can monitor latency on a number of layers, and maintains charted records of connection quality. It has a probe specific to SIP: http://os

Re: [asterisk-users] Spam

2010-11-24 Thread Steven Stromer
Same here. But, can the genie ever be put back in the bottle? > Cary Fitch wrote: >> Has anyone else noticed "new spam" in the last 2-3 weeks? >> > > No, > > But I run ASSP in front of my MTA. > > Doug -- _ -- Bandwidth and Colo

Re: [asterisk-users] One side SIP goes dead on length conversation

2009-10-02 Thread Steven Stromer
I'm under the impression that this sometimes happens when a firewall decides that the port you've opened no longer needs to be so. Are you using sip_nat? Do you have a firewall between the asterisk host and public? How are your VoIP related firewall rules configured? Has anyone seen somet

[asterisk-users] Error: Invalid SIP message - rejected , no call id

2009-07-20 Thread Steven Stromer
On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolv