Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered
On Mon, Jun 29, 2009 at 4:23 PM, James Lamanna jlama...@gmail.com wrote: On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? Yes, I've experienced the same thing. Not sure right now what Asterisk version I'm using, prob the latest in the Ubuntu 8.04 repos. Just my 2c, fwiw. -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 400 calls at g711 how much cpu power
So what happened to the OP? Seems he would be eager to help us fight the swine flu... -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
I'm currently using Ekiga. I don't think I'd reccomend it though; it lacks a lot of basic features. -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
On Mon, Jun 15, 2009 at 10:54 AM, Geoff Lane ge...@gjctech.co.uk wrote: Only received once here. Only once here also, using gmail. IOW, it looks to me like the list server had a hiccough and Christopher wrongly accused the OP. Steve did the 'accusing', not me... ;-) -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Calling Feature?
On Fri, Jun 12, 2009 at 8:43 AM, Christopher Stamper christopherstam...@gmail.com wrote: On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote: Nerdvittles.com has a nice example of this, when they are up. They used it for Phone trees for a school or something like that. Took less than 30 minutes to put in my dialplan and use Sounds like exactly what I am looking for... I went to nerdvittles.com and searched, but couldn't find it. Spoke too soon, I just found what I think you were referring to: TeleYapper. Looks excellent! Thanks! -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Calling Feature?
On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote: Nerdvittles.com has a nice example of this, when they are up. They used it for Phone trees for a school or something like that. Took less than 30 minutes to put in my dialplan and use Sounds like exactly what I am looking for... I went to nerdvittles.com and searched, but couldn't find it. you can have a script generate a list of call files which automatically ring the callers in the list and play a message I may have to end up doing that. Problem is, the people who will be recording the message want it to be really easy; like, call a number, talk and hang up. I guess I could do that, but it may end up becoming a huge project... I was hoping that someone already had done this. Thanks for the suggestions, I'll keep looking! -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic Calling Feature?
Right now, my organization is using a commercial service (OneCallNow.com), that gives telephone notifications to all numbers in a predefined list. Example: -Admin records a voice message -Service calls each number in the list, and plays the message back to them It's a pretty handy service, albeit a bit pricey. I've been wondering if Asterisk could do this for me? I don't really want to have to write scripts, but it would be great if it's already a feature. I don't have an Asterisk PBX running yet, but when I do it will probably have multiple T1 PRI lines, making it possible to dial all these numbers (100+) in a reasonable amount of time. Anyone know of a way to do this? -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe: Mute All Lines Automatically?
I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. I'm using the FreePBX web interface, and I can't find any options anywhere. Having the user mute their own line is not going to work, mosty because they *won't* do it. Also, I've used a lot of commercial web-based conferencing services. They all have lots of great features in the web interface, like a list of current participants, muting/unmuting specific lines, manual recording, dropping certain callers, etc. Assuming that Asterisk is capable of all this, is there any web-based GUI available to control meetme? Thanks! -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe: Mute All Lines Automatically?
On Mon, Jun 8, 2009 at 10:54 AM, Marc Charbonneau timebandit...@gmail.comwrote: On Mon, Jun 8, 2009 at 9:18 AM, Christopher Stamperchristopherstam...@gmail.com wrote: I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. have a look at the documentation here : http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe you want this option 'm' — set monitor only mode (Listen only, no talking) Thanks! Works great. For the record, I also found a web interface for meetme (meetme web control) that does everything I need it to. That is, one I got it to work... ;-) -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe: Mute All Lines Automatically?
On Mon, Jun 8, 2009 at 10:54 AM, Marc Charbonneau timebandit...@gmail.comwrote: On Mon, Jun 8, 2009 at 9:18 AM, Christopher Stamperchristopherstam...@gmail.com wrote: I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. have a look at the documentation here : http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe you want this option 'm' — set monitor only mode (Listen only, no talking) Thanks! Works great. For the record, I also found a web interface for meetme (meetme web control) that does everything I need it to. That is, one I got it to work... ;-) -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
