search for: earpiece

...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 hardware that do this (41D switch matrix?) But I need to find one that asterisk can use. I will then build some custom scripts to handle the "Pickup" and "Hangup" parts of it. Anyway any ideas or websites I could research for this ty...

...ndsets. 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 is the only option), no echo is cancelled and the diagnostic shows that I have clock drift. I know that the AEC expects a locked clock, but that's my current situation and I couldn't find any way to record from the same earpiece soundcard (or whatever the inte...

...it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating earpiece. Is it possible to somehow generate a ring in the earpiece until the remote phone is picked up? Bill --

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 throwing them away because they're leaking acid. Wouldn't it be a kick to be able to ring this ph...

...in this case) (2) Cisco 7940's with 7.3 firmware All communications between asterisk and the phones are over SIP. o If I call from phone to phone on the local network (100Mb switched), I hear no echo delay- sounds great. o If I disconnect the phone from the network, I can hear my voice in the earpiece (obviously, no delay). o If I check my voicemail off of the 7940 (talking directly to asterisk), I hear no delay (just audio to the earpiece from the phone itself) o If I do an echo test, I hear the delay. Here's the exerpt from my extensions.conf file that does this: exten => 10,1,Playba...

...bars on them than my phone does, and I need a bit more volume than I'm getting. 2. This phone does not act like all my others do when I am talking and a call comes in. Instead of the jarring ADSI !!!BOING!!! followed by a series of call waiting beeps, instead I get a ringing tone in the earpiece which is audible to the other party as well. Maybe there is some config setting that will handle that? Thanks in advance for any information that might be out there. B.

...nent 1 EM unit-Ear and Mouth piece, this is a headset or handset with a two position switch and a 4 conductor jack that plugs into the IP unit(ACES component 2). FOr prototyping two typical monaural PC headsets into a 2.5mm switchbox would do fine. Switch position one connects the 1st mike and earpiece to the 2 "talk" pins on the Talk/TTSControl port on the IP unit and Switch position two connects the 2nd mike and earpiece to the 2 "ttsControl" pins on the Talk/TTSControl port on the IP unit. Obviously production handsets/headsets would have only one earpiece/mike with the...

...er, but the demo systems has been purposefully configured with only basic telephony functions. Oh... someone mentioned the headset (no handset) pin jack is only for the microphone (and not the speaker) which would seem very odd. Anyone using a headset with the 942 where both the microphone and earpiece function fully? Any thoughts?

..., 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 line, I hear everything I say in my earpiece about 1/4 second after I say it. It's very irritating. We have tried 2 different * boxes, using 2 different T1/PRI cards f/ digium. After calling digium about it, we set echotraining to 800 in zapata.conf. It got better but was still there, if I turn the volume down on the phone, it doe...

...e 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 handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that?...

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 subjective, I guess. Suggestions? joe a.

Hi! I am looking to calculate the end-to-end delay between two soft phone/hard phone. I have asterisk server and configured ntp server on the same machine and synchronized it with ntp pool. I have seen that Wireshark can be used to check the jitter. But I am not sure how can i calculate the end to end. May be this is not related to the mailing list topic but please help me if anyone has some

...his is a headset or handset with a two > > position switch and a 4 conductor jack that plugs into the IP unit(ACES > > component 2). FOr prototyping two typical monaural PC headsets into a > > 2.5mm switchbox would do fine. Switch position one connects the 1st > > mike and earpiece to the 2 "talk" pins on the Talk/TTSControl port on > > the IP unit and Switch position two connects the 2nd mike and earpiece > > to the 2 "ttsControl" pins on the Talk/TTSControl port on the IP unit. > > Obviously production handsets/headsets would have only...

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

...ud) exten => 304,1,MusicOnHold(default) Here's the asterisk output from a working ATA-186 call: *CLI> -- Executing MusicOnHold("SIP/200-990d", "loud") in new stack -- Started music on hold, class 'loud', on SIP/200-990d [ beautiful music emanates from earpiece ] -- Stopped music on hold on SIP/200-990d == Spawn extension (dialout, 303, 1) exited non-zero on 'SIP/200-990d' And from the silent BudgeTone-100: -- Executing MusicOnHold("SIP/202-351f", "loud") in new stack -- Started music on hold, class 'loud...

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

...e 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 handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that?...

...u might originate a call using a SIP phone, transport that call via ISDN, but if the called end is an analog phone then a hybrid exists at that point and could create some echo. You'll find more detail on the wiki. All phones (analog and digital) feed some of your transmit audio back into the earpiece, and that is called 'sidetone'. That is sort of an intended echo. ------------------------------ Message: 8 Date: Fri, 11 Feb 2005 00:17:49 -0600 From: Rich Adamson <radamson@routers.com> Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol To: Asterisk Users Mailing List - No...