similar to: Called Party Identification

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK<<-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo

Hello, About every 2 or 3 days I notice in the messages log file: Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 2: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 3: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 4: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 5: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on

Hello, After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

Check with your telco. We had the same problem on 1 of our PRI's, every day at 5:00 sharp, red alarm, with all calls cut off for 30 seconds exactly. Turns out the equipment at the CO was going into a test loop at that time because of a forgotten setting by a tech. Man, what a finger pointing exercise that was. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent:

Hello, I have some reports from users that occasionally DTMF will stop working in voicemail and they will have to exit the system to get it to work again. The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with Ulaw codec. This is all on an internal switched 100mb lan. Has anyone else seen anything like this? Thanks, - Brent

I know this has been covered before, but could someone please explain the benefits to starting asterisk various ways. I am partly posting this too, to see if my assumptions are correct. Is call quality affected by starting it differently? My belief is no. Regardless of how you start it, quality will be the same... Correct? So far, I have used: asterisk -vvvgnc If you are logged into a

Hello, I am trying to apply the "called party identification" patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the "equivalent" of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true.

Just out of curiosity, What version of CVS and Polycom SIP software are you running happily? Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0? I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with poor results. Transferring did not work as expected. Using the # key to do blind transfers after a call was on hold did not work. Just curious. Thanks, - Brent

>>Yehavi wrote: >> > I am trying to apply the "called party identification" >> > patch (patch 8824) and managed to make it work with a >> > static data. Where do I take the name of the called person >> > (the "equivalent" of CALLERID, but the other way...)? Asnwering myself: I am using realtime extensions, so I've added call to

Hello, I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is no longer appearent on the screen when a second incoming call comes in unless I press the hold button on the first call. Does anyone have a work around for this to reject a call while continuing to talk to the first party? I should also point out that I don't want it to be on *, as the situation varies from call

Greetings, The Polycom SIP 1.5 Admin Guide says this: "3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion."

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

Has anyone had any success using more than one or two X100P cards? I have 4 in a system, and channels 2 3 and 4 all seem to work just fine. Channel 1 however is acting up. I get random red alarms, disconnects, etc. I have checked the /proc/interrupts and everything is sitting on it's own IRQ. Also checked memory addresses and everything looks good there. Not looking for anything specific,

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

We have been experiencing static noise on one of our phone lines connected to our Adtran Total Access 750 which is then connected to a T100P card. I was convinced that, after rain storms, the static would get worse, it was surely a problem of Verizon's. Verizon sent a field rep out today and he told our secretary (I was busy) that there was a hard ground on the Ring portion of the line.

Hi. I have a character vector that looks like this: > temp <- c("Company name: The first company General Manager: John Doe I > Managers: John Doe II, John Doe III","Company name: The second company > General Manager: Jane Doe I","Company name: The third company Managers: > Jane Doe II, Jane Doe III") > temp [1] "Company name: The first company

Good day Guys I forgot to add and mention a very important piece of the puzzle. We are making use of dovecots compression plugin. I.e. https://doc.dovecot.org/configuration_manual/zlib_plugin/#compression Regards Brent Clark On 2019/11/11 14:49, Brent Clark wrote: > Good day Aki > > Thanks ever so much for replying. > > Interesting that you ask the version of dovecot. Any