[asterisk-users] Answering analog call

I have a TDM400P analog card in my asterisk server. I haven't used analog for a while. The caller hears at least two rings before my 312 extension gets rang internally. Does it usually take that long? Below is my relevant dialplan. Also callerID isn't working but that might just be the test

[asterisk-users] MWI and PJSIP

Does something change with MWI when moving from SIP to PJSIP? Seems my phone isn't be alerted of its new VM. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)

Does anyone have any information for me? Welinghton. Citando Welinghton Magno Guimaraes : Hello!   I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) to make external links. Does anyone have any manual or instructions on how to

[asterisk-users] Losing my mind on MWI

Can't get MWI working with PJSIP and my Cisco phones and realtime. I have "mailboxes" populated in the endpoints and aors tables, with 312@default which is the voicemail context. I'm not sure what else to try. Please help! :) Travis --

[asterisk-users] caller id spoofing/setting on analog

I've not used analog for quite some time. It seems it's not possible in asterisk to spoof a phone number/name on an analog call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)

Frank and Kevin, Thank you for your attention. We managed to make communication work. Thank you again. Welinghton. Citando Kevin Larsen : Does anyone have any information for me? Welinghton. Citando Welinghton Magno Guimaraes

Re: [asterisk-users] Losing my mind on MWI

Ok figured it out. The Cisco needs the Mailbox ID filled in with the extension, 312, in order to get mail correctly. For some reason regular SIP works without that, but PJSIP needs that. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

Re: [asterisk-users] Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)

> > Does anyone have any information for me? > > > Welinghton. > > > > Citando Welinghton Magno Guimaraes : > Hello! > > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or