similar to: Voicemail storage

I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays "transferring" but it does nothing.

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

Hello, I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Thanks, Adam The contents of this email message and any attachments are confidential and

I just set up a Polycom 500 on *. Every few calls I make, the call connects and the receiving party can hear me (thru Broadvoice), but I still get ringing on my end, as if they never picked up. * logs look just fine. Does any one have any suggestions? Thanks. ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody

Could someone please direct me (via personal email) to a provider with good prices on Polycom Soundpoint IP 500's with POE cables? I need 14 of them. Thanks, Adam ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody Road, Suite 380 Atlanta, GA 30328 Office: 770-395-0088 x34 Fax: 770-395-0989 Mobile:

Can anyone out there please tell me what ports Speex uses? I want to set up QoS on switches but I can't seem to find this information anywhere. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended

I have a user who goes into Comedian Mail for the first time and goes thru the initial setup, changes password, records name, etc. Problem is that every time he calls in, it thinks that it's his first time and keeps reprompting him. His password change is reflected in voicemail.conf. Others do not have this problem. Where does Asterisk maintain the "first time" flag? Any ideas

I am switching from IAX2 to SIP for my inter-Asterisk transport due to assorted quality issues following the 1.2.4 upgrade. On the server that SENDS the call, I have the following in SIP.CONF: [192.168.1.2_OB] type=peer fromuser=OB host=192.168.1.2 And in EXTENSIONS.CONF exten => 91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@192.168.1.2_OB) On the RECEIVING Server in SIP.CONF: [OB] type=user

Hi Alberto, Can you please file a bug for it, with the test case attached (if you haven't already). Thanks, Ehud. On Sat, 1 Feb 2020 at 11:32 Alberto Barbaro via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Hi all, > just a gentle reminder :) Is there any update on this please? > > Happy to help in any way I can > > Alberto > > Il giorno ven 3 gen 2020 alle

Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, "In what format do

I would like to allow my remote users to dial in from their homes, cells, etc., and instruct Asterisk to forward calls made to their office extension to a number of their choosing. The wiki entry on "Asterisk call forwarding" shows how to do this. For security purposes, I would like to front-end this by asking the user to supply a password for their extension. Ideally, this would be

Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent

Hi David, In pretty sure that the crash is due to the instruction I have shared in my first email. In order to run pngpixel you just need libpng and zlib.. honest I thought that the .bc had Al the necessary. I'm happy to help you to reproduce it. As a curiosity, why do you think you cannot reproduce it? Unfortunately I won't be able to modify the interpreter soon.. anyway, I'll try to

Hi David, Thanks for the suggestions. I can definitely provide the example bc file and image ( please see the attachments ). For the debug + asserts I need a bit of more time. Anyway the full output of lli is: lli --force-interpreter examples/pngpixel_crash.bc 0 0 examples/mini.png png_ptr->width: 1 max_pixel_depth: 24 row_bytes: 28 Unknown constant pointer type! UNREACHABLE executed at

Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java

Hi, I was wondering if anyone has had a problem adding the 'delete' field to the voicemail_users table. I have no problems adding other fields e.g. alter table voicemail_users add column hidefromdir varchar(3) NOT NULL default 'no'; However when I do alter table voicemail_users add column delete varchar(3) NOT NULL default 'no'; I get a message telling me that I

> > Hey Lang - does any of this look familiar to you? I'm afraid not: I know nothing about the interpreter. As far as I'm aware it's essentially abandonware. Alberto: The usual recommendation in these circumstances is to use a JIT class instead of the interpreter. You're using -force-interpreter though, so I assume you really want to use the interpreter for your use case?

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

David, sorry for this email but I noticed I made a mistake in the previous one. So I managed to compile llvm Debug with asserts release. I have used the following commands: cmake -DLLVM_TARGETS_TO_BUILD=X86 -DLLVM_ENABLE_RTTI=ON -DLLVM_ENABLE_FFI=ON -DCMAKE_BUILD_TYPE=Debug -DLLVM_ENABLE_ASSERTIONS=ON .. cmake --build . -- -j8 && sudo cmake --build . --target install Once lli was