similar to: Outgoing call spool

We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

While the rest of you were chatting about the smallest * server, I was sitting her staring at the telephone hanging on the wall. It is a Western Electric set in a varnished pine box with an earpiece you hold in your left hand and a mouthpiece attached to the box. You crank the magneto with your right hand to signal the switchboard. The two dry cells inside are dated 1935, and I'm

ACES - Asterisk Communications Endpoint System {the following could be used by any IP-PBX but the name pays homage to Mark Spencer and friends who cannot be lauded enough for their fine work} As you read this it will be obvious I am not a professional engineer but I do have enough knowledge to be fairly certain what I'm proposing is feasible from not only an engineering, but production

I have my new phone mostly working. I do have a couple of residuals that I cannot find mentioned in the list archives: 1. Is it possible to set the volume in these things? I hope I didn't miss it, but I've looked in the doc, the FAQ, and the asterisk archives and don't find anything. The displays in the pictures all have more bars on them than my phone does, and I need a bit

Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has

When talking to me, people are complaining the volume was not high enough. The phone only allows to change the volume of the speaker/earpiece. Is there an alternative solution? Is it possible to increase the volume in asterisk? Frederic

In a system connected to a verizon T1, Digium TE411P (quad T1 echo cancellation), client is complaining it is "too quiet". The complaint regards calls over the T1, not in house SIP only calls. Their description indicates they want some earpiece feedback of themselves speaking. Also, they complain that it takes several seconds (3-4) for the other party to respond. That is kind of

Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions

Hi, (this is a minor irritation), is it possible for me to call R's editor from the R command prompt (I searched for script but that didn't yield anything). (The mouse-file-new-script route is a minor irritation )

Hi ! I have a python script that create and move .call files to /var/spool/asterisk/outgoing Sometimes...(in this case after 500 successfull calls) Asterisk don?t make the calls and the .call files are in the "outgoing" forever... Any Ideas? I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior) In my python script I move .call files using ... import shutil

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the

Hello, I'm using AEC for a VoIP application on mobile handsets. I am doing experiments to learn how to work with it, and I have a problem: As long as I play through the device's normal speaker and record using the mic, I have absolutely no clock drift (according to echo_diagnostic.m). The echo is being cancelled and all is fine. Once I route to the earpiece (and still use the mic, which

Hi All, This one has me stumped, but I've done quite a bit of debugging to hopefully isolate it down enough. Basically, I hear an echo *delay* when I do a 'echo' test (key- there's an annoying delay). Here's what I have: Asterisk 1.0.3 current as of Dec 7. x100p (although, irrelevant in this case) (2) Cisco 7940's with 7.3 firmware All communications between asterisk and

Hi all, i tried to write a call generator script. It generate i file like sample.call in ast src tree. But when i set the callerid and make a call over Modem[i4l] then the caller id is not set for the outgoing call. Is it impossible to set the callerid at this time for ttI devices? Thanks for help, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing.

Hi guys I have a lovely little irritation on my hands. I?ve been trying to set up DID-based routing in Hylafax, and I?ve come across one hell of a headache:

But it prints fine. In the printer window, under the status column, my samba printers read: Access denied, unable to connect Yet when I print to them they print fine. This is minor irritation, but I'd like to get it resolved. Does anyone have any suggestions?

Just a minor irritation. 1. When I shutdown from the GDM menu, most of the time the screen goes black and none of the "stopping..." messages are visible until just about the time the mounted filesystems are shutdown, about three lines before acpi poweroff. 2. Always when I shutdown from my desktop manager (IceWm in my case), I get all the messages immediately. Once in a blue moon, this