similar to: Asterisk in SPA9000?

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

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:

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

I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates?

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

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

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

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,

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

Hi, I'm new in asterisk world. I have questions. For example I have my server with public IP address, but two customer with softphone in a private network. How can I do to make them work with the asterisk server? Best Regards -- Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm interested in developing an automated phone survey and am curious if Asterisk could be configured to run such a system.. My idea is to record a message and a series of sub-questions. The system would call each number on a list and play the message, Depending on the touch tone response another message would be played. Is it possible for asterisk to manage a survey like this?

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

Hello, The next Asterisk Users Group meeting has been scheduled for tomorrow, January 14th at 11:30am. Its a new year and it's been two months since our last meeting. Meetings are held monthly on the second Saturday of each month, excluding July and December. Sound Choice Communcations is located in Bloomington Minnesota, just 1/2 mile west of the Mall of America. The address is: 7839

This is a system for our lab. I have no problem getting rid of X100P clones. But I am just curious why can they work. Even the drivers are not loading correctly. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent: Tuesday, May 02, 2006 10:51 AM To: 'Asterisk Users Mailing List -

Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it

Hi, I am renovating my house completely and installing new cabling for communication. I'm not to into this PBX thing but I would like to have a simple one for my house, to have different phone numbers for my family members, some kind of integration with my Broadband telephone, and possibly Skype.... Browsing the digium website didn't make me much wiser. I have a linux box that I can

Hi there, I was wondering if anyone has successfully used Asterisk as a dedicated Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or an 8 port model? Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the Mediatrix 1204 does a very good job and I figure I can build a much more

Hi all, I have a k6 2-400 that has a creative awe64 sound card. It plays sound fine when using a standard audio player. I want to use the sound card as a console phone but there is no audio from the sound card. I have tried compiling various version of asterisk with no luck. I did use this machine about 2 years ago with Asterisk and Mdk 10 and it worked fine. Currently using Mandriva 2006. Any