search 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.

Try as I might, I can not get incoming calls from ViaTalk to match against my user entry. I have both peer and user entries, and incoming and outgoing calls work, but incoming calls do not move to my in-viatalk context (they stay in the default context.) Has anyone else managed to get this to work? My user entry looks like: [viatalk-in] username=14079...

...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 what they changed (fat load of luck getting that question answered anyway). Everything was working fine with dtmfmode=inband and relaxdtmf at the default, now I've tried all 6 valid combos of those two settings with no change. This is on asterisk 1.2.27 that's bee...

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...

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 t...

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

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

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 of connections a SIP channel can handle at a given moment? I expect the line to be something simple, but I can't find it detailed on the Wiki. -- /Nick -------------- next part -------------- An HTM...

...================ It does this without caring about the fact that you are ONLY allowing ulaw in the channel configuration. I have so far played with SIP but it seems the behavior is there for other channels as well (briefly tried it on IAX as well) The problem with this is that some SIP providers (ViaTalk) only allows DTMF of the type inband, which only works on ulaw. Therefore this switch to GSM makes it impossible to enter the DISA or Authenticate password. This behavior seems to have been introduced with 1.4.12 as I didn't have any problem in 1.4.11. Has somebody else seen this. Cheers, //...

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050822/a4848b01/attachment.htm

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..... -----------...

...mple Line *************************************** [trunk0] authuser=191691245XX username=191691245XX fromuser=191691245XX secret=12345 fromdomain=richmond-1.vtnoc.net host=richmond-1.vtnoc.net dtmf=auto dtmfmode=inband insecure=port,invite qualify=yes type=peer canreinvite=yes call-limit=2 context=viatalk -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071126/f6de88e7/attachment.htm

Does anybody know what DTMF coding does S101i adapter using? I've been testing one for over a week and here are my observations: - DTMF signaling is working perfectly with Asterisk, much better than Sipura 3K Though, I think the Asterisk "iaxy" firmware is buggy, the unit is using auto-update feature; so I have Asterisk 1.2.13 and iaxy firmware version is: 23 When enable in

Hi Team, Just, I would like to know, is there any unlocked device(wireless router with built in FXS port) for home users which are connected Asterisk based VOIP service. I have looked products from Linksys and D-link etc. But all these products are bundled with VOIP Service providers (vonage, lindo and at&t) . Please sugest if any products avaiable in market. Thanks, Karun

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

Hi, Just wondering if people could suggest a good VoIP provider that can service the San Francisco Bay Area and the Los Angeles area. I've tried race.com (recommended to me) but they're kind of hard to get ahold of. Any other suggestions? This is for a business, so reliability is key. I did see the recent thread about this, and while I saw a few mentioned, I didn't see anything

It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received