search for: telecommatt

I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is there a simple way to reduce the gain without having to remix the tracks? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

...ation is being monitored, so privacy rights are not an issue. The subject will make periodic calls to the Asterisk server in order to record his movements. Does anyone have experience in this area? Thanks, Mike -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

All out of the blue I get these errors? Any Ideas why Please help May 26 09:54:28 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:30 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:33 WARNING[3964]: channel.c:507

...e Polycom, each click can be heard, the tone starts, but the tone is clipped and there is silence until the next click. I've verified that QoS is enabled in the IAD. I would appreciate your thoughts. Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

...opean server and just listen, but is there a more scientific method to collect QoS metrics? Thanks P.S. I'm getting a lot of "Page Not Found" on lists.digium.com. Are the older posts being purged? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

...ith two different Asterisk system using the same CLEC. One is * v1.2.4 and the other is v1.2.5. The systems have totally different motherboards. Has anyone had a similar problem, and what was the cause? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

I'm seeing quite a few "Not Found" pages when I google lists.digium.com. Is anyone else getting this? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

...go out to NuFone on IAX. Are these calls bridging away from the Asterisk server? How can I get accurate billing data? I tried to Google the archives but I'm still getting "page not found". Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

...th the Asterisk system or the telephones. The Asterisk version is 1.2.5. The phones are Polycom, Cisco, and Grandstream. I've checked my NIC connections and everything is full duplex. Thanks for your help. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

...ith the Asterisk system or the telephones. The Asterisk version is 1.2.5. The phones are Polycom, Cisco, and Grandstream. I've checked my NIC connections and everything is full duplex. Thanks for your help. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.

Hey everyone, I know this isn't a direct Asterisk issue, but some of you may know this answer. I recently upgraded the SIP version to 1.6.6 on all of our phones in the office. Everything is working fine, except one aspect. The phones in the office reboot randomly for no apparent reason. I haven't changed anything in the configuration files since the upgrade. The only setting in the

...t. The reason this is a big problem is that the PRI channel for the call remains busy. Subsequent inbound calls on that channel are rejected. Asterisk 1.2.12.1, Polycom SIP 1.6.6. Has anyone seen this? Thanks. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

Is there a programmatic to to trim the silence from the beginning and end of a recording? From a .wav file? From a .ulaw file? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

I currently have Asterisk 1.2.7.1 and the Sangoma A200 w/ 6 FXO ports and HW Echo canceller. I have outgoing calling setup to use a group so that if one channel is busy it goes to one of the other channels. What's weird is that when I dial an outside number, sometimes it goes through and other times I get "You have reached an invalid pager number MCLL327." I have no idea what that

Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3af cable bundle, I simply connect my phone, through the Polycom provided "special" RJ-45 cable, into a PoE capable switch, and voil?! Is this true? And if so, what happens when the Phone doesn't connect directly to the

I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386. Browsing to each device by IP address, I can get logged in using admin and I can see the advanced settings, however, if I try to change the settings and clicking the Change button, it just brings me back to ask for the password again.. I can't get into the Status page or any of the Account1-4 pages either. It just keeps