similar to: Voicemail Remote Access

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

Hi, Please take a look at the following extensions.conf:- exten => _11XXXX,1,NoCDR() exten => _11XXXX,2,Dial(SIP/${EXTEN},10) exten => _11XXXX,3,VoiceMail() I'm already using realtime for some extensions/users/voicemail. Is there any way to do the following at point 3?:- Lookup the realtime users db and read the MailBox column for that buddy. If the mailbox column is empty, play

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi, Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? Kind regards, Dan

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/076ff188/attachment.htm

Hello everyone, I successfully set up the mail_crypt plugin using folder keys, and require user's key to be encrypted with a password using mail_crypt_require_encrypted_user_key = yes. As I'm trying to streamline the process of creating a user, and want to develop an application in PHP to help me in that process, I'm very interested in the doveadm HTTP API. Although the

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? [client1_phones] include => client1_internal include => client1_outgoing_calls include => test_calls include => parkedcalls [client2_phones] include => client2_internal include =>

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hello, A couple days ago I wrote a message about replication problem with Exchange to samba-technical@: https://lists.samba.org/archive/samba-technical/2016-February/112019.html Problem I want to resolve looks like "exchange schema _not_ installed on the samba4 AD DC": https://lists.samba.org/archive/samba/2015-May/191636.html I try to search additional information and found old

Hi everyone, Does anyone have the documentation for OutCall? http://code.google.com/p/outcall/ The link isn't working. Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150

Hello, My host just had a faulty power supply and therefore, my Asterisk server was down for 7 hours. It was a Sunday so no one was making calls, however if it happened during the week, I'd have problems. I was trying to find a whitepaper or advice on how to set up two Asterisk servers to provide some redundancy. I've been googling "asterisk redundancy" but all I've found

Hello, Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work anymore. I use ODBC storage for voicemail. Comes out that the "voicemessages" table schema should have changed, because the log says Asterisk needed to store data to an additional field called "flag". Any new message cannot be saved. The thing is that I'd like to know where I can find an updated

Hi, I'm trying to view a list of the active calls to see if I can restart Asterisk. When I do 'sip show channels', I get a huge list like this (just a sample pasted):- 92.110.7.210 (None) 198827f2469 00102/00000 0x0 (nothing) No Init: OPTIONS 92.110.7.210 (None) 6b211bb04ac 00102/00000 0x0 (nothing) No Init: OPTIONS 92.108.34.153