similar to: Not answering PSTN until SIP answers

Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at

Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote

Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: > [incoming] > ; incoming calls from the FXO port are directed to this context from zapata.conf > > exten => s,1,Answer() > exten => s,2,Dial(SIP/polycom) And zapata.conf: >

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

Hello, I have asterisk server with two isdn bri cards (billion) using zaphfc driver. Also I have from telephone company routed (for example) 16 pstn numbers. It is technically possible to dial SIP phone from outside before answering isdn pstn line. I have local numbers 201,202,203 and from telecomunication company numbers 555201,555202,555203. I don't want to be "pstn"

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). I have tested the same setup with different brands of ATA's

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

I am using Asterisk and an X101P card as a glorified answering machine. We have a residential PSTN line with about six phones connected to it. Like an answering machine, I want Asterisk answer the line *only* when an incoming call is not answered after four rings. This mostly works. My extensions.conf is at the end of this message. The problem is that Asterisk will sometimes answer the line when

Hi List, I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases hosted on a separate machine). When Asterisk is in verbose mode, it prints messages saying "failed to extend from 512 to 664" (quite a few lines in a block) and then the last message is mostly "failed to extend from 512 to 663". The number of lines varies unpredictably. The full message (in the logs)

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to the PSTN network (carrier). I need to configure * to

Hi all, I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN

Hello all, taking examples from various pointers, I am attempting to put together an outbound dialing example using SIP (Cisco 7960) with 2 X100P. Everything seems to be working without generating errors, but the problem is the phone hangs up (102/Bye). Any pointers/advice are much appreciated Here is the section in extensions.conf: extensions.conf ; From CISCO at work ; exten =>

Hi! I want to get audio from the PSTN before the call is answered so I don't miss when the called phone is busy or if there is some error (like the phone is unavailable or is wrong, etc) and hear the ringing from my telco. I have polarity reversal in my telco for incoming and outgoing calls. If I set answeronpolarityswitch=yes then I get no audio until the call is answered. If I set it to

Hi guys, I've been looking on wiki, but i could find it only for chan_capi: http://www.voip-info.org/wiki/view/Asterisk+PBX+functions In the CAPI channel See Asterisk CAPI channels * Call Deflection (CD) (redirect without answering): Implemented by chan_capi How can i do it with my Softphone Xlite? Any one can help me? I want to redirect a call without answering it. Best regards,

Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000>

Has anyone had any success using the Florz patch for zaphfc ? I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN lines however the users are complaining of crackles on the line which I am assuming is related to the IRQ issues raised by Florz. I have tried to use the patch but it errors trying to patch zaphfc.h Any help would be appreciated. Regards, Stuart -- No virus