similar to: Vitelity dtmfmode=rfc2833 started working!

I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? ---- Date: Tue, 22 Jul 2008 12:23:39 -0400 From: "Mark G. Thomas" <Mark at Misty.com> Subject: [asterisk-users]

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

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

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'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test

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

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

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

I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the

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

We have an issue with incoming calls from a provider in which DTMF tones are sometimes sent using 'inband' and sometimes using 'rfc2833'. All calls are G711 and the incoming SDP never indicates support for rfc2833. Is there a setting in sip.conf that allows asterisk to receive DTMF tones in either inband or rfc2833 formats? The option 'auto' does not work for us

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 everybody, I have set dtmfmode=auto in my sip.conf, but that does not work and I still got the following message: WARNING[4980]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 According to http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode: auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones

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 simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no

We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that