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What is the best asterisk platform to use? What are you guys using? I am looking for something to host either in our data center or at the customer prem where I have the control over the unit and not through a contractor. I dont mind paying a license fee for a front end interface but still would rather not have to pay. Thanks, --Eric -------------- next part -------------- An HTML attachment

When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am hoping to get some help here with building custom PHP to manage a 'wake up call' system. I have the script where the user can set the schedule for an extension wake up call. It appears to write to the /var/spool/asterisk/outgoing/ directory. My two issues: 1 - when the files do get moved over to outgoing/ directory via a cron job, the permissions show "-rw-r--r-- 1 apache

Just wanted to get a feel for people in the community. I researched a lot of information but I wanted to know the communities perspective on this and didnt want to revive a dead thread: Any pros/cons for connection to asterisk boxes located at different sites via an IP-VPN or Frame Relay? I wanted to know the best cost effective way to connect the two remote boxes other than the Internet.

I don't think the original poster was asking about which OS is best. I think he was asking which PBX manager people are using. Ex, PBX in a Flash, Elastix, FreePBX, blah, blah, blah. Thanks; John From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Novack Sent: Wednesday, December 23, 2015 9:24 AM To: Asterisk Users

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -------------- next part

Not sure if this has been answered but I cannot find a solution. I am running Asterisk 1.4.26.3 I am seeing the following lines in my log files: A: [2013-03-05 13:54:27] NOTICE[6928] chan_sip.c: Failed to authenticate user <sip:192.210.138.12>;tag=DmVIjOlfYiiL B: [2013-03-05 12:20:00] NOTICE[6928] chan_sip.c: Failed to authenticate user 101<sip:101 at

If the end user does not pick up the phone, is there a way to log to the CDR about the call file failing? /var/log/asterisk/messages does show a NOTICE message [2015-12-31 06:58:46] NOTICE[28059] pbx_spool.c: Call failed to go through, reason (3) Remote end Ringing [2015-12-31 06:58:46] NOTICE[28059] pbx_spool.c: Queued call to SIP/102 expired without completion after 0 attempts but I would

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL: