similar to: Inbound SIP issues

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

In an effort to install cfengine (which requires 0.96b + of ssl), we've had to recompile all sorts of related packages on our RedHat 6.2 boxes. In addition, we're trying to implement an LDAP directory. Basically the source RPMS for RedHat 7.3 were installed and compiled on a 6.2 box to get this all to work. We're running into the following problem on the 6.2 boxes after having

Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jmann at txhmg.com

First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to

Hi I've 3 iax connections to my provider , each of them have own DID , PH1<----| | \/ PH2<-->|-----| <---------------------------> |----|<-- DID1 | A1 | <---------------------------> |ISP |<-- DID2 PH3<-->|-----| <---------------------------> |----|<-- DID3 I had iax phone on each of this connection , but now I want to terminate all

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

Is the IP 650 sidecar compatible with asterisk? If I pair it with the IP 650 phone, can I have more than 6 "lines" registered w/ the server? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com ________________________________ This e-mail, facsimile, or letter and any files or

Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? It's only going to support 4-5 users(the voice channels won't all be active obviously). ________________________________ This e-mail, facsimile, or letter and any files or attachments

I've been racking my brain for the last two days to try to figure out what I could possibly be doing wrong in my configuration for a SIP trunk that's setup through my local ISPs Metaswitch. I've setup a very simple SIP Peer, which I've played around with a lot in the past two days but still comes back to the following basic setup: [provider-fireball] type=friend

I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com

Hi all, I have a new installation with asterisk 1.6.1.6 but I'm unable to receive calls from a SIP trunk: [Oct 2 14:30:09] NOTICE[21554]: chan_sip.c:18523 handle_request_invite: Call from 'user001' to extension 'user001' rejected because extension not found. Are there any changes from 1.6.0 to 1.6.1 (or there is a bug)? Below my simple configuration: sip.conf

All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=5555551234 secret=password defaultuser=5555551234 insecure=port,invite context=broadvoice

Is there any way that I can have an admin user hit * and then Mute all other users in a meetme conference? Sort of a moderator function? I know it can be done with MeetMeAdmin, but as I see it that requires a separate extension to dial, unless I've got the logic wrong? If it can be done in a single extension please show examples. Thanks. ________________________________ This e-mail,

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi, Andrew. You are trying to solve two tasks: definition through what line the call came and a beautiful display of this information. 1. definition through what line the call came. If the username and password for inbound and outbound registration the same, then try the following: a) delete "register" lines. b) add option "callbackextension=Company1" to Company1 friend

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but it does work. For prosperity, the SIP service is through Internode. Here is my "extensions.conf" file: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten =>

[priv] type=friend dbsecret=dundi/secret context=longdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > > >

Is there a way to tell asterisk, via a sip.conf peer, what IP address to send a packet out of? I've got multiple NICs in my box, each with it's own public IP. I need the SIP messages to originate from any one of the IPs depending on which number was originally called(and therefore where the packet originally came from). My fear is that it will listen on all IPs fine, but only respond

I've got a couple of extensions in users.conf that have both SIP and IAX access(IAX softphone, SIP hard phone). I'd like to setup my dial string to "check" to see which they are actively registered with, and send the call appropriately. Right now I have: Exten => _4xx,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN}) But not all phones have both techs, so there is a lot of

Does anyone have a recommendation for a headset that plugs into the Mic/Line-out port on a PC? Ideally something like the Plantronics SupraPlus. I'd prefer Monaural instead of stereo, and cheap in price but not in quality. Thanks for any suggestions... Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at