similar to: Proxied SIP

Anyone else having problems with inbound Broadvoice this morning? -- Chris Tooley / Network and Development Services Networking Technologies Resource Center, LLC (NTRC) 8650 Spicewood Springs Road, Suite 105 Austin TX 78759 512-250-8985 / Fax 512-250-5909 www.ntrc.net / www.ntrcstore.com

Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can.

That's pretty much the question. I've got Symantec NAVCE running on my NT server protecting itself, the Windows clients and the Exchange server, but what do i do about my Linux based Samba server? Will a linux based AV app search for Win32 based viruses? Thanks, VeKTeReX

I've been reading chapter 3 of the SAMBA Project Documentation and I can't find any PAM files (pam.d directory) in my /etc. Is there something I need to compile in order to have these files? Thanks for any help. Darin DeCounter Atex Assistant System Administrator Las Vegas Review-Journal 702-387-5230 ddecounter@reviewjournal.com

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

What files at a bare minimum need to be in /etc/asterisk for an asterisk server that does sip only and voicemail. I'm setting up an asterisk server to provide service for a single SIP softphone extension with SIP origination and termination. The main purpose of using * is for voicemail and future expansion ability. I know I need sip.conf extensions.conf voicemail.conf but what else? do I

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

I would like to show a remark that would show call progress and appear on the CLI screen. The remark should be in the code of a sip [channel] or extentions [context] If I can't send my own remark, what little used 'show' command could I insert in the code? Can this be done? -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux

I upgraded from version 2.2.0 to 2.2.2 and since then no one can write to their home directory. Attached is a log (at 3) and smb.conf for reference. Also profiles aren't working for Windows 2000 SP2 machines. Any ideas? Thanks in advance for any help Chris Tooley -------------- next part -------------- # Samba config file created using SWAT # from 192.168.1.50 (192.168.1.50) # Date:

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd

I need to setup incoming (over an 800 number and some local DID's) and outgoing phone calls (all over the country) with an Asterisk server. This asterisk server has 20 Polycom 430 phones connecting to it. I need the best possible SIP provider out there. I have tried http://www.nufone.net and http://www.broadvoice.com and they do not even come close to the expected quality. Does AT&T

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get

I am currently using a T1 PRI from TWTelecom for DID and outgoing calls, but I recently discovered that they're offering call termination/origination over SIP trunks in my area now. If they could deliver these SIP trunks to me over a guaranteed-QoS circuit, this would be of great interest to me. We're already using a DS3 circuit from TW for our internet uplink, so I'd imagine it

Hello Asteriskers, I'm confused about why Asterisk is not a SIP proxy and why exactly this can affect the performance of a large Asterisk system. I know that Asterisk acts as a useragent endpoint, but my doubt is why exactly Asterisk could overload the call flow if the RTP voice stream goes from the caller to the called party. Does someone know how many calls or pencentaje that could handle

We use only IP connections to our asterisk boxes. Given this our origination/termination providers usually send/receive traffic to/from our network on a single IP or limited number of IPs. In a DUNDi Asterisk Cluster, would each of the boxes need to be able to connect to our origination/termination providers directly, or would we need to setup a common gateway box to forward calls to/from our

A summary of my problem, I had wrong UIDs for a user (eg pmal) between the /etc/passwd file and the smbpasswd file. When I was trying to loggin from a couple of machines running either win nt or win 98 I was reading in the log files that the loggin could not be completed because the UID were wrong but there were instances were I could loggin from another computer with the same username. I really

Well, after some other pretty horrible issues with sessionid.tdb (which I think are fixed by upgrading the version of Samba on the client to 2.2.2 as well), I have some other fun. I am now getting these messages (starting the day before yesterday): Dec 11 15:29:48 filecabinet smbd[12498]: [2001/12/11 09:29:48, 0] lib/fault.c:fault_report(40) Dec 11 15:29:48 filecabinet smbd[12498]:

When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more

I'm having a problem with files locking so that every time the are opened up they are opened as Read-Only. Windows doesn't see them as readonly but Samba thinks that they are write=no and so they can't be saved or deleted. I've included a log, my smb.conf, and a getfacl of a user's home directory as well as a getfacl of the directory from /home. Any help would be

Hello all, So I am trying to get a qemu guest to use a custom patched seabios image rather than the default /usr/share/qemu/bios.bin I have tried changing the domain config file to include either: <os> --SNIP-- <loader readonly='yes' type='rom'>/home/telemin/qemu/seabios_patched.bin</loader> </os> OR domain type='kvm'