similar to: External MWI question for Asterisk

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

Hello, I'm having troubles while using the "Goto" function in a realtime extension. Here is the error message : -- Executing Goto("SIP/siemens1-081f56b0", "script_13_0|s|1") -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context

Has anyone tried to used VB6 to communicate with the Asterisk Manager? If so, would you be willing to share some basic code showing your approach to getting connected and parsing results? I've got a Telnet control that is allowing me to connect, authenticate and see the "flow" of status, etc., but I'm sure there is a better way to do this without using Telnet (maybe not?). Any

I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context "phones" and are set to not allow reinvites. All phones can dial each other directly. The dialplan

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: <reginfo> <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.callBack="*97"

Hi list, how does one get a SNOM 200 MWI to work with * ?? When I press the MWI button it doesn't connect with voice mail on my * box. thanks

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrated via CAPI and the chan-capi channel driver into my Asterisk). MWI works just fine there. I

I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting "Explicit MWI Subscription" to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee

Hello! I faced with the following error message when I try to add a Unixware (2.1.2 i386 x86at) to windows 2000 domain (I have already added the computer to the domain), samba version 2.2.8: dab> smbpasswd -r CCS -j GEOMANT -U lhever -s Password: write_socket_data: write failure. Error = Socket operation on non-socket write_socket: Error writing 72 bytes to socket 0: ERRNO = Socket operation

Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug

I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works perfectly, however my theory is that it should be broken. Obviously I'm wrong but "Sip show subscriptions" does not show the endpoint subscribing to the MWI status on Asterisk, even though all of the other endpoints on the system DO subscribe for their respective mailboxes, including SNOM & Polycom endpoints.

Hello I am running the following setup in order to make VoIP calls at home. Home Phone <-> SPA3000 <-> Asterisk Home <- IAX2 over Internet <-> Asterisk Office The voice mail for Home Phone is hosted on the Asterisk Office machine. I wanted to have a way to check the status of my voicemail on my home phone directly. In order to do so, I've patched both Asterisk Home and

Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e1f59a92/attachment.htm>

Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David

Hi guys, is there any way to execute a command *after* a caller has hung up the call? Something like exten => s,1,Voicemail exten => s,2,AGI(mwi.agi) I'd like to turn off the MWI on my cellphone (which is done by gammu[1]) Or does anyone know a way to check the state of the MWI from outside, i.e. with a cron-job? I'm turning on the MWI with the email-notify from voicemail, but

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1. I use realtime for configuration. So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indicator lamp to blink on any of my phones (Digium, Aastra and Yealink). I have tried just the number of the mailbox and also adding the context.