similar to: Comedian Mail Wont Take Password

I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

I'm trying to figure out how to allow an extension to register more than once. For instance, I have all of these 4 line IP phones that I use with Asterisk and I would like to have a persons extension (say 101) ring at all four lines so that if the person is on the phone they can take another call, but it appears as though if you try to register the same extension more than once then the most

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

Hello, I'm having trouble getting this to work: I have a ring group that dials an extension and if no answer dials a cell phone. If the cell phone doesn't answer I want to go to voicemail or another extension. I have set the timeout to 15 seconds but it never actually works, it will just ring until the cell voice mail picks up. I'm using Asterisk@Home 2.8 and a TDM400P card.

hello I am running Samba 2.2.8 (latest) that I downloaded the binary from the Samba site. I have this problem in older versions of Samba as well. I have under Redhat 8.0 security level = user works fine under NT/2000/XP platforms, but when I try to login from a Windows 95/98/98se/Me machine, samba prompts me for a password to resource \\netbiosname\IPC$ how do I prevent this from happening in

Hi, I'm curious to know how I can programmatically differentiate between a Samba share and a Windows NT share from a Windows program. For both, GetVolumeInformation() claims the file system is "NTFS". This isn't very helpful. I need to be able to tell the two apart because "real" NTFS supports various things Samba's "simulated" NTFS doesn't, such as

Hi, I'm having a problem with the timeout option when dialing a ZAP channel. The goal is to ring a number for 15 seconds, if no one picks up, go to voicemail. The dial command is: exten => s,1,Dial(ZAP/1/6135551111,15) exten => s,2,VoiceMail(u1) exten => s,102,VoiceMail(b1) The call will continue to ring beyond 15 seconds. What's interesting is that a SIP channels does not have

Hi, I have not yet found any solution to my troubles so I though maybe someone here could give me some help. We have a Linux server with a Scsi Raid 5 disk. We use it as Samba and Nfs server for Windows and Linux clients. It also exports through Samba a disk mounted through Nfs from another Linux server. Everything was working fine until we uppgraded the server to RedHat 7.2. Now, after e few

After reading docs several times I still can't get Windows ME machines to map to my Red Hat 9 Linux machine. I can connect from the Linux machine but not to it. I'm connected through an SMC firewall/router and have tried with the firewall shut down. I can ping the Linux machine from Windows but NBTSTAT -A does not recognize the Linux box. My smb.conf file is: [global] workgroup =

Hello, I have set up Comedian Mail on my Asterisk system. I am using Voicemail not Voicemail2 in my extensions.conf file. The system works great except for 1 thing...It is not possible to create custom unavailable or greeting messages for 3/4 voicemail boxes. For some odd reason 3/4 users are unable to modify the default voicemail prompt with their own custom greeting. The greeting gets

Hi list, Is this supposed to be a joke? It doesn't sound very professional. comedian n 1: a professional performer who tells jokes and performs comical acts Moby Thesaurus words for "comedian": banana, buffoon, burlesquer, card, caricaturist, choreographer, clown, comedienne, comic, cutup, dramatist, dramatizer, dramaturge, droll, epigrammatist,

I'm trying to connect to swat but can't. Just installed Samba 2.2.2-5 and edited /etc/services, even though I'm running Redhat 7.3 and use xinetd. /etc/xinetd/swat exists.... but I still cannot connect to http://localhost:901. Why??? Thanks for assist.

Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer

A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt

I have a user who goes into Comedian Mail for the first time and goes thru the initial setup, changes password, records name, etc. Problem is that every time he calls in, it thinks that it's his first time and keeps reprompting him. His password change is reflected in voicemail.conf. Others do not have this problem. Where does Asterisk maintain the "first time" flag? Any ideas

Is there a user guide for Comedian Mail? I need to give some training materials to my end users. So far, I have been unable to find anything through google or the Digium site. THanks, Wiley The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other

Hello: We offer several options for accessing Comedian mail via Asterisk and IP phones. We'd like to add one more. We would like to be able to save a message on the Asterisk server in such a way that it can be retrieved using the telephone set and Comedian mail as it is today and/or via an IMAP client. If accessed via am IMAP client we would like to use an "externotify script" to

Hello, I was wondering if someone could tell me how I can set up a comedian mail box, and how to use it. Thank you, Hunter