similar to: ?? Vitelity dtmfmode=rfc2833 started working!

Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833" (it didn't used to), so switched my ountbound dtmfmode to

Hi All- ? I have successfully routed calls into our asterisk system from several DID providers in the USA, but for some reason I'm having a problem getting Vitelity to work. ? We are using the IAX protocol, and the symptom is that only about 50% of the calls terminate properly into my asterisk system - the rest get a busy signal.? The ones that do not come in don't show up at all on

Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom

Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

I wanted to try AsteriskNOW plus a few others to see which I can wrap my head around the quickest. The issue so far is I can't figure out how to use my Vitelity account with it. I went so far as to put their Asterisk configuration in the sip.conf file but still no joy. Any pointer as to where to search? I found a few threads in the AsteriskNOW forums and one thread from last year on

Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis

Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070324/8bb0be73/attachment.htm

Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

Hi, I'm am getting doubled DTMF on some digits with one of my providers who also uses asterisk. We're using SIP, with dtmfmode set to rfc2833, and the codec G.711. Once out of every five or ten calls, there are no problems, but more often than not, the DTMF is getting doubled-up on at least one of the digits of the extension dialed. I've tested with a CVS-HEAD from Febuary, and just

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine

I corrected my local_net setting (based on advice from network admin). I have tried several different values for the from_user and still have the same problem. Asterisk receives the OK from Vitelity. Asterisk sends the ACK (without a Contact header). Vitelity doesn?t seem to process it, so they send an OK again. The OK receive, Transmit ACK occurs 4 times. A short while later, Vitelity hangs up

I am not sure if I entered the correct settings for the transport information. For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered. One minor detail, we are using ip authentication. When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for

Yes, everything is behind the same NAT. For the application I?m working on, the only endpoint is the endpoint to Vitelity. We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones. After that, we control the call through AMI to perform the work we need. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of

Thanks George. I will correct my local_net in the morning. Vitelity chan_sip settings I have working, do not have a fromuser. sip.conf settings... [HVout] type=friend dtmfmode=auto host=64.2.142.93 disallow=all allow=ulaw canreinvite=no trustrpid=yes sendrpid=yes nat=yes context=TestApp On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph at fairview5.com<mailto:george.joseph at

Yes, outbound calls are the only ones I?m trying. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph Sent: Monday, December 15, 2014 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at

I am running PJPROJECT 2.3 and Asterisk 13.0.0. I answer the call, about 15 seconds later, vitality hangs up on my cell phone. However, Asterisk is never notified When the OK (for the answer) occurs, the ACK seems to never be accepted. The OK recvd with ACK sent occurs several times. Here are the pjsip.conf settings... [global] type = global debug = yes [transport1] type = transport bind =

Trying this again after my first away from work in a couple weeks. Running Asterisk 13.0.0 IP authentication with Vitelity I can Originate with sip, but not pjsip. Here is the sip settings and trace. Action: Originate ActionID: S8 Channel: SIP/8005555555 at outbound.vitelity.net Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable: