similar to: Specifying DID for outbound calls

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

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201' - but that gives everything and parsing that isn't really an option. Anyone know how

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. 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

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

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,

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 have a problem that has developed within about the past 3 months with my backup outgoing SIP provider (I am not sure when this problem started since it involves only my backup provider which is used rarely). The problem is that most (not all) outgoing calls fail during the earliest stages of call setup, specifically after the provider sends back a "407 Proxy Auth Required" response.

Is anyone else using Vitelity right now and having an issue with a global outage of sorts? Potral/WWW arent accessible and it would appear through monitoring that the outbound is flapipng like mad. The outbound can be rerouted, I know, but inbound is a huge problem right now. [Sep 4 10:26:13] NOTICE[27507]: chan_sip.c:15679 sip_poke_noanswer: Peer 'vitel-outbound' is now UNREACHABLE!

Hi, I've been trying to use the FaxGateway application to send T.38 out over Zaptel using asterisk but I don't seem to be having any luck. I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number]) Has anyone had any luck using this thing and can enlighten me on how it's supposed to be used? Thanks.

Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-2222). The problem is that this extension was

I am able to return users and groups using wbinfo -g and -u. Samaba will even allow users to connect that are in our domain. The problem exist while trying to narrow down permissions to a share. [public] comment = Public Stuff path = /home/ public = yes read only = no valid users = @"UFAD\_IFAS-FRE-USERS_autoGS" This does not work. It prompts the end user for a

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

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