similar to: SIP RTP flow

Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden

Version: platform = sparc-sun-solaris2.6 arch = sparc os = solaris2.6 system = sparc, solaris2.6 status = status.rev = 0 major = 0 minor = 64.2 year = 1999 month = July day = 3 language = R -- It is unclear to me that the handling of NAs is desirable, and it has problems with names: > z <- c(T,T,NA,F,T) > names(z) <- letters[1:5] > which(z) Error: names attribute

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

Hello, I'm preparing a setup before installing it within the next few days. In this setup, I'm using a SPA112 as an ATA for an analog phone. The target phone is a Gigaset A400 DECT handset. In my lab, I've got another A400 handset and an old Matracom 46 handset. When I connect my Matracom 46 handset to my SPA112, I can send and receive calls. When I connect my A400 handset to the

Anyone have practical experience using inexpensive GAP-compliant DECT handsets with the Snom M3 basestation? When I asked Snom support, the answer was that 'basic functionality should work', but they didn't elaborate. I'm _guessing_ that means registering/unregistering with the base, making calls, and receiving calls (including presenting caller ID). They also stated that they

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

No kidding, every time. I know I have the config via tftp working. Funny story - I was getting nowhere with it and then decided to tcpdump on the tftpd box, and wow! The UIP-200 tftp client was looking for the uniden<mac>.txt in lower-case! Hah! That was easy to fix. Now the config is transferred to the UIP-200 at startup. It registers to the * server. The phone displays time and

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I have come across the above handset a few times in the UK, They are quite cheap over here (~?80) Not the best handset in the world but works well enough. I have been asked to setup a central config server for a large collection of these handsets. I know they can do Auto provisioning via FTP/HTTP/TFTP I have got an example of the generic

Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram.

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't

We have calls that originate click-to-dial apps that use the manager interface. As most of you know these apps first ring your handset so that you pickup the handset and then place the outbound call once you have picked up. When they first ring my handset (before me picking up the handset) the call shows as being "from asterisk". Is there any way to change this "from" name to

After 4 hours of debugging codecs, changing config files, etc. as a result of not being able to capture voice from a Cisco 7960, I eventually found that the mic in the handset appears to be dead. Does anyone know where I can get a new handset (just the part you hold to your head, everything else on the phone works fine)? Or, does anyone know how to open one up? I tried doing a little prying

Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit "R" key on

Hi list! I'm having some problems with chan_sccp and a Kirk IP600. Basically the handsets work (they emulate a Cisco 7940) but I have the following issues: 1. If a handset is in a conversation and there is a new incoming call, the incoming audio is muted (but the other party can still hear anything spoken on the handset). What is normal Asterisk behaviour, that a handset is left alone

Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". The echo I mean is when I hear myself while talking to another party. I have heard

More an echo algorithm question than a purely asterisk one... I have the following setup: Handset - PAP2 - Asterisk - SPA3000 - Telco And no matter what I do, I get echo on a call routed out via the PSTN when I talk into the handset, in the order of a hundred ms (my estimate, could be wildly inaccurate!). Echo will occur also when I have a handset plugged into the phone port on the SPA3000