# Checking voice mail remotely?

**URL:** <https://community.asterisk.org/t/checking-voice-mail-remotely/2567>\
**Category:** Asterisk Support\
**Created:** [November 18, 2005, 2:20pm UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567 "2005-11-18T14:20:42Z")\
**Posts on this page:** 7\
**Page:** 1

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**Author:** ![randy998](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/randy998/32/1207_2.png) [@randy998](https://community.asterisk.org/u/randy998)\
**Post date:** [November 18, 2005, 2:20pm UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567/1 "2005-11-18T14:20:42Z")

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how do you check your voicemail remotely with Asterisk@Home 1.3? Or should I ask can you check remotely? We normally set up systems so the last trunk is the backdoor to the voicemail just for this purpose.

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**Author:** ![netoguy](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/netoguy/32/1258_2.png) [@netoguy](https://community.asterisk.org/u/netoguy)\
**Post date:** [November 21, 2005, 2:23am UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567/2 "2005-11-21T02:23:20Z")

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i’m not really familiar with \*@home… if you have an extra DID, the easiest thing would be to set one to route to your voicemailbox. alternatively, you could simply set things up so that dialing a # would reroute you there on your regular DID.

i’m a little fuzzy on exactly what you’d add to your extensions.conf in order to make this happen, but i know it can be done.

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**Author:** ![rightbrain](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/rightbrain/32/511_2.png) [@rightbrain](https://community.asterisk.org/u/rightbrain)\
**Post date:** [November 21, 2005, 3:59am UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567/3 "2005-11-21T03:59:33Z")

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[quote=“derek”]  
i’m a little fuzzy on exactly what you’d add to your extensions.conf in order to make this happen, but i know it can be done.[/quote]

```
    exten => 500,1,VoiceMailMain()

```

Dialing ext. 500 at the main IVR prompt drops the outside caller into the VM system.

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**Author:** ![sshoyat](https://avatars.discourse-cdn.com/v4/letter/s/d26b3c/32.png) [@sshoyat](https://community.asterisk.org/u/sshoyat)\
**Post date:** [December 3, 2005, 3:48am UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567/4 "2005-12-03T03:48:17Z")

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I was able to access the comedian voicemail by dialing \*98 and entered a mailbox number and password but for whatever reason it keeps saying password incorrect. I can log into comedian web with the extenion and password and it works perfectly (i.e. ext 100 password 100).

Thanks.

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**Author:** ![dickson](https://avatars.discourse-cdn.com/v4/letter/d/57b2e6/32.png) [@dickson](https://community.asterisk.org/u/dickson)\
**Post date:** [December 3, 2005, 3:58am UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567/5 "2005-12-03T03:58:08Z")

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Not sure if this was over kill, but i pointed a DID to a queue in @home, and simply had the queue dial \*98 each time a call entered it.  
Seems to be working pretty good.

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**Author:** ![sshoyat](https://avatars.discourse-cdn.com/v4/letter/s/d26b3c/32.png) [@sshoyat](https://community.asterisk.org/u/sshoyat)\
**Post date:** [December 4, 2005, 6:14pm UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567/6 "2005-12-04T18:14:46Z")

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Ok, everything seems good now. I had to make a modification to my grandstream device…

Send DTMF: in-audio via RTP (RFC2833) via SIP INFO

I changed it from in-audio to via RTP (RFC2833)

Once I made those changes I was able to dial \*98 and get into my voicemail with no problems.

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**Author:** ![zmanea](https://avatars.discourse-cdn.com/v4/letter/z/8dc957/32.png) [@zmanea](https://community.asterisk.org/u/zmanea)\
**Post date:** [December 4, 2005, 6:43pm UTC](https://community.asterisk.org/t/checking-voice-mail-remotely/2567/7 "2005-12-04T18:43:07Z")

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Good to hear that you solved it. For future reference, should anybody ever have this problem and find this page, there is no need to use a queue to access Voicemail main. A@H configuration is a little different then plain vanilla Asterisk, but in Asterisk you can configure an extension that goes straight to Voicemail main. Something like:

exten =\> \*98,1,VoicemailMain(${CALLERIDNUM})  
exten =\> \*98,2,Hangup
