search for: digitalpath

...dom 30 second "pauses", no IO can happen and things get really backed up on the system and it ends up being faster for me to just dump the tmpfs completely and go back to writing directly to disk as slow as that was... Any suggestions? -- Ray Van Dolson Linux/Unix Systems Administrator DigitalPath Networks

I would like to setup some rules for a wireless bridge to do load balancing. I''d like to dedicate one radio for doing transmit and one for receive. router | | ------------- eth0 ---------------- | | wlan0 upstream device wlan1 | ^ |

...t on yet another network. I'm hoping to get all SIP clients (PAP2-NA's) to register at the central Asterisk server so I can avoid setting up a whole bunch of Asterisk servers within each private network. Here is a generalized diagram of how I hope things could be layed out: http://webdev.digitalpath.net/~rayvd/voice/voice-network.png I figure if the load on our main Asterisk servers gets too high it would be fairly trivial to add additional servers to cope with this. I'm trying to figure out what options I have for "proxying" the connections from the PAP2-NA's directly to t...

On a custom compiled Linux 2.6.13 kernel... # tc qdisc add dev ppp145 handle ffff: ingress # tc filter add dev ppp145 parent ffff: protocol ip prio 50 u32 match ip src 0.0.0.0/0 police rate 384kbit burst 10k drop flowid :1 RTNETLINK answers: Invalid argument This works fine on a CentOS machine and my Fedora Core 2 box with default kernel. I''m trying to figure out what is missing in the

Does anyone have any recommendations for network monitoring systems that work with CentOS 3 or 4? thanks

We have a small app that runs asterisk -rx "sip show peers" (etc) and gathers some info and reports it back somewhere else. This is called from a cron job essentially and doesn't really run all that often (once a minute). We noticed some gaps in the results we were getting and dug in a little further. We found that if you run asterisk -rx "sip show peers" from the

Disclaimer: Yes, I know faxing over G711 is unreliable. :-) We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway. We're running Sipura SPA-2002's as ATA's and faxing within our own voice network is working. If we try and fax out to the world however, we're running into a problem. When the call connects and the modem tones begin to negotiate, our SIP/PSTN

I've been having problems getting MusicOnHold to work, so I've dumbed down my setup to as simple of a setup as I can. Asterisk 1.0.9. SIP ATA's (Sipura SPA-2002's) <SIP ATA 1> <---> <Asterisk> <---> <SIP ATA 2> Both ATA's have public IP's. No NAT'ing going on here. Reinvites are allowed so the media stream bypases Asterisk once a call

When qualify is set to yes in sip.conf for a "friend" and the OPTIONS packet gets returned with an ICMP port unreachable message, what is the behavior of Asterisk? It looks to me like Asterisk tries sending the OPTION request again right away (well within a second or two). Some of our devices are being Linux firewalls that make use of iptables to do portforwarding. This generates

I'm running into an issue where subscribers to our service cannot call certain 1-800 numbers if they have a caller id blocked account (restrictcid=yes). This is on Asterisk 1.0.9 and our clients are using Sipura SPA-2002's. Our provider uses a SIP/PSTN gateway, so we hand off SIP to them from Asterisk. The problem appears to be that when an "Anonymous" call goes out, there is

I'm trying to figure out how Asterisk decides how often it will send SIP NOTIFY's to an ATA when a voicemail message is waiting for the user on the server. >From watching, it seems to be completely random. Sometimes 10 seconds apart, then 33 seconds, then 13 seconds, etc. Each time causes a "ring" on the ATA. I can change the ATA's behavior, but what I'd really

I''m trying to set up customer egress shaping on concentrator that customers connect to using pptp (which uses pppd). This means customers each have a pppXXX interface when they connect. It''s easy to shape their "down" speed this way, and when they disconnect the interface is gone and thus the shaping rules automatically go away as well. However, shaping their upload

Our setup: SIP IAX2 SIP <SIP Phone> <-----> <Asterisk1> <------> <Asterisk2> <-----> <PacWest SIP> | RTP \-----------> <PacWest RTP> I can make or receive calls from my SIP Phone, however voice only works in one

Posted a similar version of this to the RHEL4 list... I''m having trouble getting CentOS 4.2 installed on a DL140G2 with two SATA drives. I am doing a PXE install, so I do not have a CD-ROM drive in the system and this appears to have the side-effect that both SATA drives are caught by the Uniform Multi-Platform E-IDE driver before ata_piix can catch it. As soon as I hook an IDE CD-ROM