search for: vitel

...[global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>* [outbound.vitelity.net] type = aor remove_existing = yes qualify_frequency = 60 contact = sip:64.2.142.93 [outbound.vitelity.net] type = endpoint context = TestApp transport = transport1 aors = outbound.vitelity.net dtmf_mode = rfc4733 force_rport = yes rtp_symmetric = yes rewrite_contact = yes send_rpid = yes tr...

Hi, all. I'm probably doing Something Dumb(tm), so please feel free to point out whatever I'm missing, no matter how stupid. Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get: Rejected connect attempt from 64.2.142.19, who was trying to reach '6031234567@' This leads me to my first question -- why doesn't it show a context? (My second is, what's wrong with the snippets, below?): iax.conf: [vitelity] context=viteli...

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 full configurations that are available are a bit complex. Any tips or recommendatio...

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...

On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at amtelco.com> wrote: > > 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. > > > > And it's outbound calls that aren't working right? > > > > > *From:* as...

...etwork admin so he can verify the settings I entered. > > > You need the network and mask. For example if the ip address and mask of the test machine is 192.168.0.1/255.255.255.0 then the correct entry would be 192.168.0.0/24. > 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 the account. > > > > Should these settings work without the from_user (IP based authentication) > or do I need to get the account name from Vitelity? > >...

...at? > > > Just to be clear...both the pbx and local endpoints are behind the same NAT? > [global] > > type = global > > debug = yes > > > > [transport1] > > type = transport > > bind = 0.0.0.0 > > protocol = udp > > > > [outbound.vitelity.net] > > type = aor > > remove_existing = yes > > qualify_frequency = 60 > > contact = sip:64.2.142.93 > > > > [outbound.vitelity.net] > > type = endpoint > > context = TestApp > > transport = transport1 > > aors = outbound.vitelity.net...

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the sip.conf they sent me, everyt...

On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com> wrote: > > 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... > > I think you can actually specify anything, it just has to be populated with something other than a sub-account username. > [HVout] > > type=friend > > dtmfmode=auto > > host=64.2...

...e gateway. Is there anything special I should to to make this work? Note my soft phone does not have any issues using the same dialing rules and extension information. Here is some of my config stuff: ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116 5060 Unmonitored 101/101 68.156.63.118 D N 1038 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]...

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

...n update... My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have. He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net At this point, it seems to be working (and this is going through a Cisco ALG). I will run more tests, but here is the pjsip.conf I have. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net] type = aor remove_existing = y...

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 on my cell phone. Asterisk is never told the call is gone. If I hangup the call from Asterisk...

...ler ID ;exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=XXXXXXXXXX) ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _NXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound) exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXNXXXXXX,n,Set(CALLERID(name)=&qu...

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 the account. Should these settings work without the from_user (IP based authentication) or do I need to get the account name from Vitelity? Have a great day! Da From: asterisk-user...

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 respon...

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 George Joseph Sent: Mond...

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 fairview...

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 supposed...

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 up >> sending a reinvite which their side & they do not support us sending a >> reinvite. Ive tried...