similar to: Discriminating between two incoming lines?

I'm using a modified "default config" file for extensions.conf, the one that uses macro-stdexten to handle the stations. We use a TDM30 card for our stations. When a call that has been rung in using that macro transfers the call things work just fine as far as the "other" instrument ringing. But once the ring timeout has expired, the call then drops into the *original

I'm sure there must be a setting I'm missing somewhere, so I thought I might was well ask here. Conversations are punctuated by sudden replacement of a given syllable or so of conversation with a DTMF tone. I would hope perhaps there's some kind of setting that has to do with the way it detects inband DTMF? I'm pretty sure it's an artifact of this particular ATA; my

I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of the bug tracker tickets that it is instigated by a small network leg looks to be on point to me,

I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

Thanks for the reply Bill. I am aware of the interrupts problem. To solve it I have already disabled my serial ports freeing up interrupts 3 and 4 and these are allocated to the two cards. This was done 2 months ago and has not solved the problem. Is there any way that something can wake up every now and then and generate these two interrupts? My current /proc/interrupts is as follows:

I am interested in running asterisks and gno phone. However, i have yet to purchase the x100p cards. Is there a way for me to test asterisks in like a simulated mode or something? (to get used to it and the configs and stuff, prior to purchasing the x100ps?) Thanks all, A.J.

Anyone here have experience with these devices? They would ppear to be an affordable alternative to multiple X100Ps. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "Kick at the darkness 'till it bleeds daylight" - Bruce

Hello folks, I'm very interested in the Digium/Asterisk combination but need some clarification. I would like to setup a SOHO for business and home use. Scenario One: I have one analog line, 4 analog telephones. Do I need a TDM400P + 4 FXS modules (Green) + X100P? Scenario Two: 2 analog lines, 1 selective ring number for fax, 8 analog phones. Is this what I need? 2 TDM400Ps and 8 FXS

I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike

All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing

Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 Til:

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

List members, Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS

I wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

I wonder if anyone could send me a pointer to technical specs and pricing information. I got a mail today from an acquaintence that contains what I believe is some serious misinformation, referring to the 7960 as their new portable 802.11b SIP phone. A quick search of eBay would seem to refute that. I hope this is an OK question to ask. . . Thx. b.