similar to: asterisk voicemail retrieval using a cisco 7940

While setting my first couple IP phones, I set their voicemail buttons to an extension that runs VoicemailMain. exten => 8500,1,Wait(1) ; voicemail exten => 8500,2,VoicemailMain ; exten => 8500,3,Hangup ; I would like to be able to pass the mailbox number allowing each phone to go in directly but I'd rather tno have

; Below assumes you are using the same number for Voicemail boxes as extensions ; if ${RDNIS} is blank then GotoIf will go to extension 2, otherwise it will go to extension 102 exten => 8500,1,GoToIf($[X${RDNIS} = X]?2:102) exten => 8500,2,VoiceMailMain(s${CALLERIDNUM}) exten => 8500,3,Hangup exten => 8500,102,VoiceMail(u${RDNIS}) exten => 8500,103,Hangup ; you should now be able

Hey Guy! I want direct access of VoiceMail without asking mailbox number (Direct ask PIN). I am using following dialplan but its still asking me Mailbox number. Look like asterisk 1.8 doesn't support CALLERIDNUM variable. Do you have any idea ? exten => 8500,1,answer exten => 8500,2,wait(1) exten => 8500,3,voicemailmain(${CALLERIDNUM:-4}@default) exten => 8500,4,hangup exten

I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hello list, I want to listen to voicemails on my * box from a phone that is not local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm aware that I can forward VM to email or use a web interface but that is not always practical. Other than doing an IVR type arrangement or a phone number dedicated to VM access is there a way to do this? On my old POTS line I used to be

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823

Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/c9e420c5/attachment.htm -------------- next part -------------- Hello all, I have 1 IP 7940 with the following Firmware versions App Load ID: P00303011201 Boot Load ID: PCO303010001 Version 3.1(12.1) Could you please confirm, if my IP phone has the correct SIP image. My asterisk

Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but

I have some Cisco 7905 phones with the SIP load 1.02.00(040406A). When the phone is off-hook but no call has been placed, or when the Do Not Disturb is activated, the phone returns a 302 "Moved Temporarily" message back to asterisk as follows: ----------- -- Executing Dial("SIP/5060-0811bb00", "SIP/9871234|20|Ttr") in new stack -- Called 9871234 -- Got SIP response

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

Has anyone had any major issues with the Cisco 7940 and or 7960 phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030818/2c156d1f/attachment.htm

I want to buy a 7940 to use with Asterisk. Does all the features (i.e. Transfer, Hold, call waiting, MWI, etc)work? How difficult it is to configure 7940? Thanks __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/

Hi All. After some input, I created a V1.1 version of my ODBC VM retrieval from the ODBC_Storage It now uses either Mysql or unixODBC drivers to connect to the database I didn't have php compiled with unixODBC so i had to recompile it in "./configure --with-unixODBC --with-mysql --with-apxs2=./blah/blah/blah" make make install after all that it worked. see the Readme and

Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). No new features - all bug fixes according to the release notes. I've already started using it. I thought those of you running the Cisco phones and the appropriate access who didn't yet know would like to

This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of

Ok. I was sucessful in installing ODBC storage I'm using MySQL in the backend as it is. but all my drivers are now ODBC. I am running asterisk-cvs head as of last night 9/19/05 My question is this... the old voicemail.cgi script that allowed checking voicemail no longer works etc, and never did work for me without a static voicemail.conf file. Anyways.. that aside... how does one retrieve