similar to: how to block spammer calls

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Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

Hello, Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? In particular, with a hardware configuration similar to: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) I have two fully independent systems

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ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why

Sorry, I'm doing a mail test since I was not able to send any mails to the mailing list for about a week... Thanks,

We have the caller id with name option enabled with our provider, however, our polycom 501 phones will only display the number of the incoming call. Is there a way to see the callerid name from the cli when the call is coming in (like a print in the dial plan)? I'm not sure if the problem is with asterisk or our phones. I did turn on the calleridpres option in zapata, but I'm unsure what

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring (starts sounding normal, then sounds "weird" -- like two rings played at

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev

Hi all, This is my first post to the list. I have searched the net far and wide but can't find an answer to this problem. When I have call forward working or use the voicemail from a SIP phone, the first part of the message is always cut off. So instead of hearing "call forward cancelled" I hear "l forward cancelled". Or in voicemail I hear "edian mail"

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Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

I have always been able to block toll-free numbers by catching them with a line similar to this for each DID I have on my system: exten => 5554441212/_888NXXXXXX,n,Playback(GoAway) Where 15554441212 is one of the DIDs that rings into our Asterisk box. The problem with this approach that I have to create a line like this for every pattern I want to block multiplied by every DID on my system,

I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue with it. We can call 911 which is routed out these new trunks, and it goes to the 911 center, but they are not getting the ANI and hence "no record found". Our LEC is Embarq, and they say they can see the call come in and send: KP-911-ST and then KP-0-911-ST rather then KP-0-ANI-ST I turned on all the debug

We have a PRI from Telepacific. Asterisk 1.2 and a Sangoma A101 T1 card. Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

Hello, I just got off the phone with my PRI provider, and was told that I am not sending an expected message when I reject a call with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how