similar to: Calls not going through

I get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Feb 19 21:14:35

Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call

Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to

Hi list, I was wondering if anyone had any solution to either one of two issues I'm having: I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, it works very well for the most part, but after less then a week of heavy usage, eventually the phone gets into a state where it won't accept or let you place any more calls, the screen flashes "no free lines

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

Hi list, This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges

Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio

We are testing our Asterisk server prior to deployment. The server has a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and one PRI for local calls. We are using sipp from two different stations routing a test number out the LD lines and another test number out the PRI line. We can not get more then 23 total active calls to connect to the test numbers, the test numbers

I have 2 sip phones and an analog phone attached to a Digium USB fxs device. I would like the analog phone to ring when transfers are made to it, but I don't want it to ring when a call comes in from outside, although I would like the person at that phone to be able to pick up the phone and answer the incoming call. Is that possible? Thanks Sean Garland, MCP+I, A+ Siskiyou Technology

Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a

Hi. I''m loving mocha but have ran into a problem with using the mocha plugin with a project that has the SslRequirement plugin. It seems there is some conflict between the two? Any ideas? $ ruby test/functional/calendar_controller_test.rb /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.0/lib/active_support/dependencies.rb:478:in `const_missing'': uninitialized constant

I have a very annoying problem that we hear on our end, but the other party doesn't hear. There are random beeps and echo type noises that occur. They are present during voicemails, and present on my end during calls. Is anyone experiencing the same deal? I have asked this a number of ways on the list, and never get a response... Thank you. Sean Garland Mount Shasta, CA

It appears that my replies aren't getting to the list. Just testing to see what is going on... Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/cc18a485/attachment.htm

I have no experience with GIS whatsoever. I really know squat about it. I'm currently working on a project to plot locality data for wildlife on a map of Shasta County. This is for a new herpetological club that hopes to track strength of our local herp populations, since it seems evident that in the last 20 years we lost one species of frog and are close to losing another (probably less

We are experiencing a delay when using the MeetMe service. There is a 1/2 second of delay heard by all parties on the conference call. This delay seems to be consistent with all connected parties, meaning that if there are four connections to the conference room (A, B, C, & D) when user 'A' talks users 'B', 'C', & 'D' will all hear the audio with

I just bought an I Audio mobile device (Btw, these thigns are the coolest players on the face of the earth.. might want to check em out). It plays ogg vorbis files rather nicely. My whole music library is sitting in flac format. When I encoded my flac files (the majority a year ago) I used replay gain to save the album and track information if the flac file. Unfortunately, the I audio doesnt like

I have been looking around some and I can't seem to find anything which will answer my question. If I have two Asterisk boxes in different locations which are linked to each other over the internet, can I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line

While I am still in the process of working through this here are the steps I am taking to set up each machine. 1) Since each of my servers have multiple NIC ports I bond them together (see page 34 of the Administrators Guide) first. This is best down from the physical server and not via the remote console. If you do it remotely you will then have to reconfigure the management port before

Hi everyone, I'm having a bit of an issue after upgrading from asterisk ~1.2.24 to 1.6.2.11, with the old version when the user would go to transfer a call, they would press Transfer, then the speed dial button for the extension, optionally introduce the call, and then press Transfer again to complete the transfer. Now, with the new version, when you hit the speed dial button, asterisk

On Tue, 2004-02-03 at 14:30, Joshua M. Brown wrote: > Would someone pleeeeeeease put a filter on this list to stop all the > nonsense messages my inbox keeps getting hit with from this list? > > thanks. > > j > I second that. It's getting insane.