Displaying 20 results from an estimated 6000 matches similar to: "Polycom 601 - To not make noise when there is VM"
2006 Dec 27
3
Polycom 601 Contacts List
Good morning,
I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in
2006 Nov 29
3
Polycom 601 Second Incoming Call
Hi List,
I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times
2014 Oct 14
1
Issue playing high quality white noise
Hi,
I have a client that wants a phone system that will play sounds from a
sleep machine. I tried using all different formats (GSM, WAV, WV49,
MP3 etc.). Over SIP it was OK however with the PSTN it broke up from
time to time. I assume this has to do with the fact that the PSTN is
limited to 8khz. Is there something I am missing here or is this
simply a limitation of the PSTN?
Regards,
2007 Nov 28
1
Polycom MWI's will not turn off
Hello,
I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that
the MWI indicators will never go off (The blinking red light and envelope in
the LCD).
I have tried to upgrade to 1.4.14 and all different SIP versions on the
Polycoms. I am now at 1.6.7
Here is the SIP Message that turns on the lights:
Scheduling destruction of SIP dialog '
2006 Oct 19
2
Polycom boot error
I am having the same issue as below. Has this issue been solved or
does anyone know an answer? This error recently began and we have
multiple phones out of commission. PLEASE HELP!!
http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html
How did you find out about 468*??? It's sure as poop not documented in
the Polycom Admin Guide anywhere.
-----Original Message-----
2006 Dec 07
7
Running Asterisk on a Home rotuer
Hi list,
Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ?
Thanks.
Dovid
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2006 Nov 12
3
Determine if Call is from a cellular phone
Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US.
Thanks.
Dovid
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2007 Jan 09
2
Attatching VM via email for more than one user
Hi List,
I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI:
uniqueid => 17
customer_id => 0
context => techmast
mailbox => 14
password => 1234
fullname => Sales and Service
email => user1@domain.com
email =>
2007 Sep 18
3
Interesting Conference Request - Can this be done ?
Hi List,
I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to).
Any ideas ?
Thanks.
Dovid
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2008 Jun 01
5
New faxing protocol. Good/Bad ?
Hi List,
I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and
2007 Apr 26
2
Changing Voice from Male to Female
Hi List,
I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa.
Thanks.
Dovid
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2007 Nov 26
1
OT: Best firmware for Linksys Router that is "SIP AWARE"
Hi,
I am currently playing with DD-WRT and I like it. I am looking for something that is more "SIP Aware". Anyone know one those that are out there ?
Thanks.
Dovid
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2006 Oct 23
2
Digium vs. Sangoma
I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ?
[02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave.
[02:15] <Dovid> u serious ?
[02:15] *** mog
2006 Jan 22
1
Gen. Question
<RANT>
Funny your concerned about copyrights and moral issues regarding the
work of others.
One question you may want to ask YOURSELF is:
Why would I use as my email a copyrighted work followed by the
name of the Company that owns the copyright???
asteriskdigum@yahoo.com, Come on!! Who are you trying to fool? Are you
out for the fast buck, by having someone that thinks you work for
2019 Jan 15
2
MWI Delayed on Polycom VVX phones
Hi all,
When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes.
After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing.
Installing Asterisk 13.24.1 did not fix this.
Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in
2007 Dec 21
3
Polycom 330 beep on new VM
Hi,
I have a Polycom 330 that emits a beep every 30s or so when there is a
message waiting. Is there a way to disable that? It is pretty annoying.
Regards,
Ugo
2006 Dec 03
1
RTP Media Path
I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer.
1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP
2007 Jun 21
1
AudioCodes Gateway and Asterisk
Hi List,
I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 "Unsupported Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ?
Thanks.
Dovid
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2007 Feb 27
2
Polycom Firmware
Hi Guys,
A while back (several months ago) I was having issues with wmy Polycom's and Asterisk. I was told to use a specific set of firmware and sip version. I am unable to find that email. Anyone know which ones work well with Asterisk ? (I believe it was 2.x )
Thanks,
Dovid
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2019 Apr 19
2
Forking AGI or GoSub
In PHP something like:
$pid = pcntl_fork();
if ($pid != 0) {
// we are the parent
// do parent stuff
exit;
}
// we are the child, detatch from terminal
$sid = posix_setsid();
if ($sid < 0) {
die;
}
// do child stuff
On 04/19/2019 02:00 PM, Mark Wiater wrote:
> On 4/19/2019 1:49 PM, Dovid Bender wrote:
>> Mark,
>>
>> I am using PHP agi and when forking