Re: [asterisk-users] T.38 w/ MAX TNT ASTERISK
What TAOS do you have? On May 21, 2008, at 10:39 AM, JR Richardson wrote: We solved our former echo issues... however, as luck would have it..Faxing is yet a completely different animal. We understand that digium doesn't really support faxing with asterisk... HOWEVER...it seems that ATA manufacturers and the MAX TNT indicate that fax is supported Scenario: TDM/PRIs --MAX TNT -sip- asterisk -sip- ata -fax We should note that the network between the ata and the facility with the max and asterisk is a managed fiber network in a type of campus environment... 1-2ms latency end to end tops... If the ata is reinviting to the MAX TNT shouldn't fax work with T. 38... Does anyone have any experience with this configuration ? Thanks, I have been wanting to do this for months, but just can't find the time to work on it. If you do get it going, I would really appriciate knowing how. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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 ___ -- 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] Coppercom and Asterisk
Mike, I might be able to help you. Contact me offline. s-h-a-n-e-b-AT-m-e- t-r-o-s-t-a-t-.-.n-e-t Shane On Mar 4, 2008, at 1:12 PM, Mike Hammett wrote: I was doing it because of the volume on the server. It is very easy to miss a message or 10 or 100 on a list of this traffic. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 10:30 AM Subject: Re: [asterisk-users] Coppercom and Asterisk On Tuesday 04 March 2008 09:45:38 Mike Hammett wrote: *bump* If people don't know, they don't know. There is no need to repost your query 10 days later. Not that many more people have signed up, and those who have signed up are unlikely to be able to answer your question. The only thing that this does is serve to annoy the rest of the people on the list. Please do not do it again. -- Tilghman ___ -- 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 ___ -- 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 ___ -- 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] Call Limits
I have the sample problem. Just turned it off for now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remi Quezada Sent: Friday, August 17, 2007 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Limits I think its an Asterisk bug, call-limits stopped working for me once I upgraded from 1.2.16 to 1.2.18. There is a bug opened for it, but the issue hasn't been resolved yet. Here is the link: http://bugs.digium.com/view.php?id=9794 -Remi Ira wrote: At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --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 ___ --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 ___ --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] DTMF intermittent on menu.
Anyone seen DTMF control lost intermittently inside a menu? We have had a few problems with one particular menu but both seem identical but maybe different traffic. Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sample PRI and FXS channel bank zap files for zaptel and asterisk.
Anyone taking a PRI and turning it into FXS channel bank channels? The spanmap and trunk map has me confused. Id like a TG of PRI and a TG of FXS channels on my FXS channel banks. Anyone have an example? TIA! Shane ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Single T1 card with Echo Cancellation to work withDell?
Don't know about the single T1 but the a104d works flawlessly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warren Sent: Wednesday, June 14, 2006 1:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Single T1 card with Echo Cancellation to work withDell? HI all, I was on this list back in Dec-Jan but the asterisk server got pushed back in the project queue and it seems to have finally risen to the top. I am looking to deploy * running on Centos 4 on a Dell 2850. I need a single T-1 (PRI) card with HW echo cancellation. I had been told that the digium cards were having problems with Dell servers back in January and also told that Sangoma was due to have a single T1 with echo cancellation out by March at the latest (by an email from Digium themselves). I was wondering if anyone could give me a heads-up on the state of single T-1 with HW EC cards. Thanks, W ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FreePBX virtualization
We have a revision of this that we use in house. We are interested in working with others on a version 2 skipping some of the mistakes of our first version and using a better model. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX virtualization Any alternate open-source solutions? On May 25, 2006, at 2:17 PM, Douglas Garstang wrote: Yes, but it fast becomes a provisioning and management nightmare. -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Thursday, May 25, 2006 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FreePBX virtualization You can by creating different contexts and using the Administrators function allow them to modify some of the settings themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 10:42 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] FreePBX virtualization Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speaking of echo canceling...
I've seen very small echo on two boxes connected with IAX2 and SIP phones. Polycom 500's. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Grey Sent: Sunday, October 30, 2005 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Speaking of echo canceling... Can I please get a definitive answer on this from the list... I have read through lots of material, and I am 90% sure the answer is no, but the occasional offhand statement has left this little nugget of hope in my mind. Is there any echo cancelation that I can enable or add for pure IPIP calls? I frequently get bad echo on IAX2/ulaw connections between my CVS HEAD and a friend's 1.2 Beta, using analog phones connected through IAXys on both ends. Neither PBX has a zap interface. We understand that this is primarily a problem with the acoustic/electric properties of the telephones we are using, exacerbated in conditions of higher latency, but is there a software canceler that I can apply to the problem? Replacing the devices is not really an option. (Although, just for information, is there a list somewhere of standard telephones known to have exceptional acoustic and signal isolation between earpiece and mic? He is using some Uniden cordless sets that are otherwise very nice and were certainly not cheap, so apparently price/overall apparent quality is not necessarily an indication of favorable de-coupling characteristics.) Thanks for your help. lyd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo canceller on TE406 Asterisk
I've had similar problems with no fix. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Thursday, October 27, 2005 10:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Echo canceller on TE406 Asterisk Hi, I have TE406P (2nd gen card with echo cancellation on-board). We still notice quite often echo on our PBX that is connected to one of the spans on TE406P (with calls routers to PRI provider on another span). I've tried to experiment with the echo cancellation on asterisk. I enabled echo cancellation in Zapata.conf to see if I can improve the situation and users started reporting warping bubble (description I got from one of the users) sound on calls from PABX-Asterisk-PRI (and other way). I was expecting that asterisk would disable its echo cancellation once it find on-board module. The strange thing I noticed that after system reboot things are now better. Although I cannot say for sure because the system was ever rebooted 2 times. Can anyone shed some light on this? Has anyone had similar problems? Or point me into right direction for troubleshooting? Regards Boris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Opinions on IAX JitterBuffer in old-school 1.0.0?
I've had similar problems with IAX2 and ticks. Jit buffer on and off don't seem to change things much. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, October 27, 2005 7:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Opinions on IAX JitterBuffer in old-school 1.0.0? I have a small issue with some remote users connecting to my primary Asterisk server using 1.0 Every few seconds, there is a subtle tick and a very small amount of jitter. The tick is not consistent i.e. it could be in 2 seconds, could be 5, could be 10. This does not affect core functionality, and the call is quite usable but my endusers want it eliminated. My primary is also running 1.0. I am using the AstShape script on both ends and tos is set to 0x18 on both ends. If I connect through IAX to my primary with a 1.0.9 ([EMAIL PROTECTED] 1.5) there is no tick. GSM codec end-to-end, no transcoding. SNOM SIP phones on the remote, latest firmware. CPU on both ends is basically nil. No IRQ conflict. The tick only exhibits itself through my firewall but note if I use 1.0.9 though the firewall there is no tick. If I bring the remote IAX box inside the LAN and plug it in, no problem. Default jitter buffer settings on both ends as follows: jitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 Roundabout average-type values for IAX2 SHOW CHANNELS is 20ms lag, 10-20ms jitter, 60-80ms jitter buffer My theory is somehow the jitter buffer is contributing to this. Before I start twiddling knobs, I'd like opinions on whether the jitterbuffer could be contributing to this, and whether fooling around with the jitterbuffer values would have any effect. For various reasons, I am staying with 1.0 for the moment. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension availability
Any info on getting the SNOM 360 and * working with DSS buttons? Regards, Shane From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Monday, September 26, 2005 2:29 PM To: Joshua Laroff; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension availabilty The snom360 phones along with the current CVS-HEAD of Asterisk can presently do this. You'll want to do a wiki search on Hint in the dialplan for implementation details. Polycom has also just released a DSS sidecar to go with their 601 model phones, but the firmware to support more than 8 appearances at a time is still in the works. If you need something now, I'd go with snom360's and Asterisk. I have deployed this already in production and it is working quite well. The DSS LED lights solid when the person is on the line, and blinks when their phone is ringing with an incoming call. On 9/26/05, Joshua Laroff [EMAIL PROTECTED] wrote: I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated. Thanks in advance, -Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE406P audio drops
Are the drops in one direction or the whole call? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Tuesday, September 06, 2005 11:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE406P audio drops Hello, Now that we've had our new Digium TE406P card in production for 4 days we have discovered audio drop problems that happen randomly across all channels. Here's more about our setup: P4-3.2GHz 2GB ram Slackware Linux 10.1 with custom kernel 2.4.29 Asterisk 1.2beta1 Digium TE406P quad T1 card with the following attached: - 2 x RBS D4/AMI 24 channel T1s - 1 x RBS B8ZS/ESF 24 channel channelbank - 1 X PRI B8ZS/ESF crossover to another asterisk server The audio path is entirely T1 channel to T1 channel. all calls in and out go to meetme conferences. This exact setup had no audio drops last week when it had a TE405Pv1 in it. The audio drops happen on different channels for up to a second at a time. We had about 100 of these small audio drops yesterday randomly across a total of 14,000 calls. Any help fixing this would be appreciated, MATT--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Working NFAS config w 411p anyone?
I finally figured out that echo directives and channel specific stuff needs to go between group and channel otherwise it didn't work or just gave weird results. I still have a problem with fax detection in terms of it turning off echo canceling. I have tried both, incoming, and everything in between. Seems like a way to turn it off (echo can) in the dial plan would be useful rather than having to answer the call with fax detect. Thanks, Shane -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Friday, August 26, 2005 12:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Working NFAS config w 411p anyone? Shane Burrell wrote: Does anyone have a working NFAS config for Zapata and zaptel for 2 NFAS trunks? First two DS1s on tg 1 and other two on tg2? I just setup two TE411Ps a few weeks ago, each with an NFAS group on it (one was two spans, the other three). I followed the documentation in the sample zapata.conf file and it worked fine... when I remembered that the 'logical span numbers' start at zero, not one :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working NFAS config w 411p anyone?
Does anyone have a working NFAS config for Zapata and zaptel for 2 NFAS trunks? First two DS1s on tg 1 and other two on tg2? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any one using the new Digium echocancellationcards
I have experienced similar problems for the past few weeks with no resolve. Last message I received I was the only one with the problems but now I see it starting to be more prevalent. In my case the card is unusable and has left a bad taste in everyone's mouth around here. SB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard A. Smith Sent: Wednesday, August 17, 2005 8:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Any one using the new Digium echocancellationcards David, Yes we got them and they caused huge problems. The echo training would cause the line to mute and you would hear something like a dtmf tone briefly and then you would be connected and talking again. This might happen once or 50 times during a call. I spoke to Digium and they say there may be a firmware upgrade coming down the line. You are right it is a daughter board placed in the space provided on the existing cards. We disabled the daughter board and are basically back to where we started from. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Zanetti Sent: Wednesday, August 17, 2005 4:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any one using the new Digium echocancellation cards On Wed, 2005-08-17 at 08:56 -0500, Alan Bunch wrote: THe wiki doesn't seem to have any user reports. If your using them, how are the working, better, worse about the same. Also what hardware seems to be stable with them installed. I'd also be interested if the module is available as an upgrade to existing quad boards. It looks rather like the echo canceller is a daughter board, connected roughly where there's a connector on existing quad boards... Have a rather nastry PRI echo problem which so far no fiddling with settings (tx, rx, taps) has helped with. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel Problems with 1.0.9
We are having similar problems with the studder. Eventually we get a kernel panic or the lines just get to the point it studders every few seconds. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Traue, Jr. Sent: Wednesday, July 27, 2005 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Zaptel Problems with 1.0.9 I'm experiencing rather severe problems with 1.0.9 (we've had to backrev to our last version we know works (1.0.5). We are running a single PRI line with a T100P card. After about 10 hours of asterisk running and the modules loaded we start hearing noise and stuttering on any call that passes over the PRI line. I've tried this with echo cancellation on and off with no difference. This is a new problem for us as 1.0.5 behaves perfectly in this regard (it has it own issues, but that's another story). We would like to move back to 1.0.9 however restarting out phone system (which is in production) every 10 hours isn't really an option. Is anyone experiencing any similar symptoms, and if not what information would the developers need to work on this. Please note that running unstable isn't an option as the only PRI line I have to play with at the moment is our main line. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Grandstream phones.
Anyone with any comments on DSS buttons and general phone features? Thanks, Shane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] US$200 bounty for * paging feature
Paging already works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Litwiller Sent: Wednesday, April 20, 2005 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] US$200 bounty for * paging feature Henry, I'd certainly be interested in your paging code. Henry Devito wrote: I am already doing this with AGI, PERL, and PHP to set up the page groups. I will release the code as open source if people are interested. I'm not the best PERL scripter in the world but it works. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Request for PRI Dump
I'd be glad to provide this. Where and what do you need? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Wednesday, February 23, 2005 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Request for PRI Dump Hello, To assist Matt's efforts in bug 3554 (2BCT CNAM), I'm hoping someone can provide a dump of the setup and related messages from a PBX that supports outgoing Station Name to the CO. As suggested in the bug, I tried to ask my telco for a dump of the setup messages for a client that supports this but was told to contact my vendor as they cannot provide that information. Obviously my vendor is this community. Any help would be much appreciated! Thanks, Trev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as a voicemail for a central office switch
* can be used in a CO switch. As long as you can do DTMF interface. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Waterman Sent: Wednesday, February 23, 2005 8:46 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk as a voicemail for a central office switch I've spent the past several weeks reading up and playing around with Asterisk while I've been waiting for an ISDN card I got on ebay to arrive so I can really get to business. I'd just like to run my project ideaa by some of you to hopefully get a little feedback. I aplogize if this ends up being a somewhat long message. In the Marine Corps we've somewhat recently started using Redcom IGX switches in place of the proprietary (and ridiculously expensive) switches that we've used in the past (well, and still do use). This opens the door to all kinds of off-the-shelf equipment that we can interface with. My goal now is to usefully integrate Asterisk primarily to provide voicemail and autoattendant feature and probably to demonstrate various VOIP capabilities. Ideally I would interface via a T1/E1 interface (might as well use E1 for the extra channels) but since I don't want to shell out gobs of money for what is at this point still a personal project, I'm opting for ISDN connectivity. One interface for now will give me two channels but I'll get another interface when I see more progress. To my knowledge, the Redcom switch isn't specifically designed to be interfaced with any voicemail systems. However, it can be set up to forward calls on busy/no answer. This looks like it should be the ticket. I can set a forward number for each phoneto something like yyy where will just tell the switch to follow a specific route to a group of ISDN channels. Then it'll open an ISDN connection and pass on yyy (and whatever else is needed) to Asterisk which could be used to identify the voicemail box somehow. I guess I could set up any type of extension (SIP or IAX or whatever) and since nobody would ever actually be connected to that extension, Asterisk would automatically send it to voicemail every time. I imagine autoattendant could be set up similarly. My understanding of Asterisk is very.. limited at this point as I'm sure you can tell. But does my above thinking seem doable? I imagine I'll figure it all out eventually but any pointers that might speed up the process would be greatly appreciated! Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
