similar to: Voicemailmain() refusing connection problem

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

Sorry if this hit the list twice, sent out yesterday, but didn't see it show up. Hi All, Can Asterisk be used as a SIP proxy, blah, blah, blah??? I've glanced over questions like this through the years, with a good idea on what a SIP proxy is and what Asterisk is and IS NOT. I never really took the time to lab-up SER and test drive it to see what advantages might be gained from using

I have a extension 981 setup for entering VoiceMailMain: exten => 981,1,VoiceMailMain,([mailbox]@usvm) exten => 981,2,HangUp() I want to pass the calling extension to the context (extension and mailbox numbers are the same). This dosen't seem to work. I get this in the console: Asterisk Ready. *CLI> -- Executing VoiceMailMain("SIP/2504-ba66",

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

This is a SER/Polycom question, but I hoped we may have some SER guru's here... I have a series of Polycom phones that are tying to register with OpenSER. The phone sends a REGISTER message, and OpenSER replies with Unauthorised (all normal). The phone re-sends the REGISTER with the credentials, and OpenSER sends Ok. Here's where it goes downhill. The polycom's appearance display

Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based proxy / call routing setup? I need to get simple CDRs; not for detailed settlement/rating, but just for reconciliation with an ultimate TDM carrier just to make sure we only get billed for what we're actually using. I'd use the often-heralded approach of dumping a call from OpenSER into Asterisk and having it

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hello guys Is there anyway for me to change the sounds that are presented in VoicemailMain? For instance, instead of it saying "mailbox", I would like it to say something like "please enter your mailbox number now". Is there a way for me to do this? I also noticed that when in some of the menus, even if I select one of the announced options it simply repeats the same menu

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

Hi everybody, I have corrected this line in extensions.conf by stripping spaces off and now it executes: exten => 22999,1,VoiceMailMain(s${CALLERIDNUM}) when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number. Anybody knows why? Thank to you all, very kind members of this list! Ciao Mauro

Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/98a6f962/attachment.htm

Hello, I'm experiencing a weird problem when using the VoiceMailMain application. If I use the application after dialing a Local channel, there's strange beep just after asterisk answers the call and before the first locution. The extensions.conf I'm using is: Ruido extra?o al llamar a la aplicaci?n VoiceMailMain [default] exten => _X.,1,Dial(Local/${EXTEN}@test) [test] exten

Hi in asterisk 1.4, I need to cancel the password check and allow users enter in the mailbox without entering password. I tried this: exten => 911119,1,Set(LANGUAGE()=es) exten => 911119,n,VoicemailMain(${Mailbox}@default,s) exten => 911119,n,Hangup and this: exten => 911119,1,Set(LANGUAGE()=es) exten => 911119,2,VoicemailMain(s) exten => 911119,n,Hangup But it does not work,

Is there a menu tree diagram somewhere for the Voicemailmain application? I know my users will ask for one, and before I started drawing my own I thought I'd see if someone already had. --- David Brodbeck, System Administrator InterClean Equipment, Inc. 3939 Bestech Drive Suite B Ypsilanti, MI 48197 (734) 975-2967 x221 (734) 975-1646 (fax)

Hi All, The situation: When I dial into VoiceMailMain(@context), put in my VM # 1001 and Password 1001, no problem, but at the voicemail main audio prompt (Alison), when I ?press 3 for advanced options? then ?press 5 to leave a message? I put in a mailbox number 1002 within the same [context], but VoiceMailMain looks for the mailbox in the [default] context and will not recognize the mailbox I?m

We have been working a OpenSer/Asterisk solution to replace our Avaya PBXs.The OpenSer is to provide scalability and the Asterisk to provide rich features.I know this has been many times for calling card platforms but I'm not sure if anyone has a good scalable solution they are using on their virtual PBX or in a CPE PBX environment?If so I would like to talk to them about buy their deploying,

Dear Asterisk Users, I'm a Japanese and now configuring Voicemail. Now I need to modify the way cmd VoicemailMain works to fix language difference and other my conveniences. What I want to do are... 1) Add words used in message retrieving guidance. I need to add different suffixes to numeric words due to Japanese way of mentioning time. (e.g. in English, you can say "Five