search for: huntbrothers

.... If I don't add anything to the sip.conf file for the phone, I can dial internal extentions, but when I try to add entries for it to the sip.conf file, it gives this error : Jun 2 14:07:44 NOTICE[-1116451920]: chan_sip.c:6642 handle_request: Registration from '<sip:5kevin2@asterisk.huntbrothers.com>' failed for '209.16.88.183' Jun 2 14:07:57 NOTICE[-1116451920]: chan_sip.c:6334 handle_request: Failed to authenticate user <sip:5kevin2@asterisk.huntbrothers.com>;tag=43325296 Jun 2 14:07:58 NOTICE[-1116451920]: chan_sip.c:6334 handle_request: Failed to authenticate use...

...<asterisk-users@lists.digium.com> Message-ID: <FE3AABFA37FDE645A491291B1FDF1E2B2BF503@hunt02.hbcorporate.com> Content-Type: text/plain; charset="us-ascii" can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com

You are correct... No glare on a PRI W. Kevin Hunt CCIE #11841 www.huntbrothers.com -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steven Critchfield Sent: Tuesday, May 25, 2004 3:26 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] "Glare" condition - How well do...

...2 months and getting no takers (and very few if any people asking about it), we are almost finished writing it in house. I'll keep the bounty up untill we do finish our product so if anyone beats us to getting it working they'll get paid... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > C Wegrzyn > Sent: Tuesday, September 28, 2004 5:56 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject...

I am trying to hook up my Cisco telephones to Asterisk using a Layer 3 switch and am having difficulties it getting it to work. I realize this may not be the proper forum for a discussion on VLAN architecture and configuration so I won't post the question here. I though I had read all the requisite information regarding the configuration for this, but perhaps I am missing something simple.

We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS

An interesting article for those needing ammunition to sell Asterisk within their organisation or to others: "Is open source IP telephony ready for prime time? Yes" by Zenas Hutcheson, St. Paul Venture Capital Network World, 06/07/04 http://www.nwfusion.com/columnists/2004/0607faceoffyes.html On a related note, they also have an article arguing the contrary position (see link within

can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Christopher Jacob > Sent: Saturday, September 25, 2004 9:51 PM > To: asterisk-users@lists.digium.com > Subject: Cisco Downloa...

Has anyone tried to get dialogic ss7 trunking to work with Asterisk? I did some googling but nothing helpful turned up... Could a person get an intel dialogic and get * to see it like a zap channel? And use it for incoming and outgoing trunk access? -- respectfully, Joseph ------=============

Dear Asterisk Community, I would like to offer my most humble apology for the enormous stream of mail messages currently battering at your mailboxes. I was honestly trying to help a worthy cause. Please blame me and only me. Neither Russell nor Josh are at all to blame for this. Nor, for that matter, is Olle. I am the guilty party. Please direct all anger/outrage/hatred/etc. towards

Hello! I wanted to make sure that, in addition to my complaints, I make it very clear: Digium's support is excellent. The jury is still out on the usefulness of the TDM products. However, Digium has worked very hard to make sure that this issue is resolved. I actually got an e-mail from someone at Digium actually asking what they could do to make me happy! She even gave me

Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider?

Hi- I have an upcoming application that requires use of PRI channels that are primarily used for high-volume incoming traffic, but that are to be used for outbound calling as well. Of course, one option is to have dedicated outbound channels reserved, but this is an inefficient use of channel resources. Normally PBX's are designed to have the CPE yield to an incoming call if a particular

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

Hello, After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't

...hotraining=yes echocancelwhenbridged=yes jitterbuffers=5 channel => 1-23 group = 1 language=en signalling=fxo_ks musiconhold=default echocancel=yes echotraining=yes echocancelwhenbridged=yes jitterbuffers=5 context=default channel => 97-100 W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com

Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a

Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR, the list goes on and on. We want to replace this with several asterisk boxes each running one or two 4 port PRI cards. The problem is: I can plug in 20 PRI lines into the CSX (from PSTN) and have 1 come from CSX into asterisk. If 1 call comes in on each of the 20 pris,

Hi everybody, I will receive my CISCO 7960G tomorrow. I've ordered it as a "global spare" without any callmanager licence. Now I don't know if I can get firmware-updates so could please someone send me the SIP-firmware? Is the default firmware the "skinny" one? Wich would be better to use with asterisk? Thank you very much

Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am!