similar to: sip.conf user entry for ViaTalk

Is anyone else with ViaTalk experiencing an outage right now? My DID has been down since 5AM (8/20). Asterisk is unable to re-register or connect for outbound calls. I have also tried calling support and their number gives a fast busy.

I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

I just put in a new Asterisk@Home 2.8 system. Trunk is connected via SIP to ViaTalk. I had an older Asterisk@Home system up and running that was working fine and I replicated settings over to the new box. When I call 7777 from an internal SIP extension I can hear the IVR menu just fine. However, when I call from a POTS phone to our number and it comes in via ViaTalk over SIP the call connects

Hello, I have a Linux machine auto-mounting an NFS share, then sharing it out via Samba (not my idea). Everything is fine - except: On Windows machines that have mapped a drive to the Samba share the Total Size and Free Space for the mapped network drive (Samba/NFSshare) shows Total Size of 20.0 MB and Free Space of 0 bytes. To rule out a very simple Samba problem I created a new (local)

Since I can't seem to get anything figured out for the Comedian system, are there any other systems out there that we can hook asterisk into? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/32ae9d25/attachment.htm

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

Hi Guys, I have noticed a weird behavior in 1.4.12. When using Authenticate or DISA in the dial plan the channel immediately switches to gsm format (if you request a password) or slim (if you run DISA without password). The debug log says... =============================== [Oct 14 21:23:00] DEBUG[9013] channel.c: Set channel SIP/1970xxxxxx-0821aad0 to write format gsm [Oct 14 21:23:00]

if you have anyone questions about your service you can contact us at the support 978-418-7300 James Jones Broadvoice Technical Support ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Ben Wern Sent: Sat 8/28/2004 4:34 PM To: Asterisk Users Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting Can anyone who is using Asterisk

since delete is a reserved word, what do you name a column in your voicemail options table to allow setting of the delete option for realtime voicemail? Anyone? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sorry, sent with wrong account....read below _____ From: Sherwood McGowan [mailto:sherwood@viatalk.com] Sent: Tuesday, August 23, 2005 8:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: SIP DEADLOCK Anyone using a CVS-HEAD pulled later than 8/13? We're runnign a downloaded CVS-HEAD from 8/13/2005 and getting SIP Deadlocks like crazy.....

It gets hard to read my logs when every time someone makes a phone call it displays long pages of "Dropping voice frame". Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ************************************** Asterisk Standard debug (level 3)

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco

I can give you a suggestion that worked for me for the after reboot no printer access. Make sure your printing daemon starts before your smb starts. It seems that samba won't see an lpq/lpstat if there isn't one. Funny how that works!! Hope this helps, and if I'm way off base, someone let me know... Wade -----Original Message----- From: epn.neustadt@t-online.de

First of all, I completely understand your frustration. It is (unfortunately) a very common thing in IT project. IMO, this sounds like a problem with project methodology and communication to me. There is a secondary concern with the adoption of a new technology after the start of the project, but I think the main problem is method. You may want to research an agile approach like Extreme

Does anyone on-list know of a serice provider that can provide DIDs in 713-861-xxxx? I'd like to port my AT&T POTS lines to an IP based service into my Asterisk box. Michael

ok, i cant make i crash. but i notice my cpu load. 17535 username 20 0 294852 13716 11520 R 100.0 1.3 280:28.70 winbindd 542 username 20 0 291040 16720 14172 R 99.6 1.6 281:45.88 winbindd and the funny thing.. this user isnt logged in anymore since the last 6 hours. Greetz, Louis >-----Oorspronkelijk bericht----- >Van: samba [mailto:samba-bounces at

Hi all I have a box with Centos 4 and qmail with all pathches. since Last month i have had twice the same problem. The process kswapd0 uses all my resources of CPU and RAM memory and the consequence is that my local queue grow up and the box go slow too. The first time i guess the problem was the version of the kernel, so i did an upgrade form kernel.i686 2.6.9-5.0.3.EL to kernel.i686

i found the following also .. Aug 7 08:44:10 rtd-print1 autofs[15991]: Starting automount.... Aug 7 08:48:26 rtd-print1 autofs[16291]: Stopping automount.... Aug 7 08:48:27 rtd-print1 automount[16302]: syntax error in map near [ * server.internal.domain.tld: ] Aug 7 08:48:27 rtd-print1 autofs[16297]: Starting automount.... Aug 7 08:48:46 rtd-print1 rpcbind: rpcbind terminating on signal.