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Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

...came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

...y using a hardware deskphone. It seems simple - but nothing so far fits the bill. Can you recommend something? Thanks! Milen The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. Compuware Limited (company number 1522537) is a company registered in England and Wales whose registered office is at 163 Bath Road, Slough SL1 4AA, Berkshire...

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...advice as to what release I should upgrade? Or should I just stick to 1.4.x and just upgrade DAHDI? Thanks. Regards, John Lee The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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...____ This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies.

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...it mean that after DAHDI is installed, we should not yum update kernel because it will affect the operation of DAHDI? Thanks. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140512/5472ff8...

I remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks. Doug.

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

...ng and outgoing calls from SIP to ISDN. Is this setup in any way supported by *? Regards, Hans Vledder The Netherlands -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060

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Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a >reload extension is performed, those dynamically created extensions in the regcontext vanish. Now

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...stinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.