RE: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
If you're up for it, I've done this a few times before , and with asterisk. Contact me offlist, and I can help. - Original Message - From: Brandon mailto:[EMAIL PROTECTED] Comouche To: asterisk-users@lists.digium.com Sent: Monday, March 12, 2007 6:11 PM Subject: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone - hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers :-) Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy _ ___ --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] Open CallerID Database?
You MUST account for fraud, as well. Perhaps proving you own the number, as in the LNP process, by providing the cover page of the bill... _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion Sent: Monday, February 19, 2007 3:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Open CallerID Database? On 2/19/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. YES! Would creating a public database, managed by users be worthwhile to anyone? I'm not sure the technical issues will be as easy to work out as one would hope. When creating such a system, care must be taken to keep the information accurate and up-to-date. And where would you get the information from in the first place? Thanks - Any input is greatly appreciated. What I would like to see is a distributed system that allows for updates to be rsync'd in, so that those of us who keep our servers off the Internet can move it through a QA process and then push the update through. Some type of a mirror system, where the packages can be updated from time to time (like daily). -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com http://www.xstreamhost.com/ - Web Hosting http://www.SophMedia.com http://www.sophmedia.com/ - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://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] How to exit from console?
you have to start it with no options in order to -r into and quit out of it _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, January 23, 2007 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to exit from console? I personally run asterisk in a screen session. Gets rid of this problem and makes things a lot easier. - Original Message - From: Marco mailto:[EMAIL PROTECTED] Mouta To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion Sent: Tuesday, January 23, 2007 1:41 PM Subject: Re: [asterisk-users] How to exit from console? Try safe_asterisk , for an easy way to start asterisk in background, and then connect with asterisk process running asterisk -rx Now you can use exit, and by the way you may look on wiki diferent ways to run asterisk. On 1/23/07, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --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] Asterisk On FreeBSD
yep. email me offlist. I can help you. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid Exeplish Sent: Wednesday, November 22, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk On FreeBSD Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task ___ --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] Sticky Polycom 501 keys and handset
I had this EXACT same problem, and 2.0.x is the problem according to Polycom Tech Support. I had such a hard time explaining the problem, too Downgraded to 1.6.7 and all worked well again. Polycom says if youre using Asterisk, dont go past 1.6.7 until they say to. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Tuesday, November 07, 2006 11:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sticky Polycom 501 keys and handset Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I'venoticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number and the key comes up (the hardware seems fine) but the phone produces this lng tone as if I had pressed the key for 3 seconds. Even the receiver is sticky, giving my dialtone when I lift it only1-2 seconds after I lift the handset. It simply looks like the phone can't keep up, like a sluggishcomputer. Anybody has ever seem this? I'd like to downgrde to SIP 1.6.7 to see if the new sip app was the problem. How can I do that? I've placed the old sip.ld file where I had to, but the phone wont pick it up. Short of that, can somebody point me to the newest firmware (2.0.2) to see if thatwould help? Mike ___ --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] Sticky Polycom 501 keys and handset
hmm, Id like to know that. How do you reboot remotely ? J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Tuesday, November 07, 2006 2:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sticky Polycom 501 keys and handset Disregard my previous message, I succeeded in downgrading my phones. And it worked, thanks Rick for the info. Is there any Polycom-specific mailing list I should be on to be aware of stuff like that? Also, would you know how to check the version of sip.ld remotely? I know how to reboot remotely, and I did for a few phones, but my paranoid self would like to double check and see if the sip.ld 1.6.7 re-installed ok by checking the current version. Is that even possible? Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Smith Sent: November 7, 2006 11:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sticky Polycom 501 keys and handset I had this EXACT same problem, and 2.0.x is the problem according to Polycom Tech Support. I had such a hard time explaining the problem, too Downgraded to 1.6.7 and all worked well again. Polycom says if youre using Asterisk, dont go past 1.6.7 until they say to. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Tuesday, November 07, 2006 11:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sticky Polycom 501 keys and handset Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I'venoticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number and the key comes up (the hardware seems fine) but the phone produces this lng tone as if I had pressed the key for 3 seconds. Even the receiver is sticky, giving my dialtone when I lift it only1-2 seconds after I lift the handset. It simply looks like the phone can't keep up, like a sluggishcomputer. Anybody has ever seem this? I'd like to downgrde to SIP 1.6.7 to see if the new sip app was the problem. How can I do that? I've placed the old sip.ld file where I had to, but the phone wont pick it up. Short of that, can somebody point me to the newest firmware (2.0.2) to see if thatwould help? Mike ___ --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] Queue Status via Dialplan
Using queues here (1 of them), and would like to know if anyone's written anything like a script that might tell someone by festival or the like of the status of a queue, like # of calls waiting, and hold times... Any other way of finding that out without spending a ton of money on third party packages ? R ___ --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] Running Multiple Instances of Asterisk
you didn't listen. SIP only. Anyone can understand that multiple instances on the same machine can't touch the same hardware. I can see how this would be very easy - dedicate an IP to an instance, and it'll play nice. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Monday, September 25, 2006 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Best of luck getting multiple instances of Asterisk to play nice when accessing Zap channels. James Texter wrote: Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --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] how to transfer a caller out of a queue ?
can't the agent just transfer the caller to another extension, whether that be another queue or a person ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan-Michael. Guenther (in-put GbR) Sent: Monday, September 18, 2006 3:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to transfer a caller out of a queue ? Hi, I would like to give a caller the chance to leave a queue after an agent has already accepted the call. The caller enters the queue by dialing 333: [from-sip] exten = 300,1,Answer() exten = 300,2,Queue(q1|tT) When the caller presses # and e.g. 1, asterisk is looking for this extension in the context where the call came in. In my configuration this means, that my office phone is ringing: exten = 1,1,Answer() exten = 1,2,DIAL(CAPI/@8304499:8304498,30,tTr) exten = 1,3,Hangup But in this case not the caller, but the agent has been transferred! Isn't there a chance for the caller to stop the conversation e.g. because the agent told him that he has called the wrong queue and that he should dial #1 to get to the right queue or directly to another person? If the agent does this, the caller get's transfered to the office phone, as expected. As far as I understand the documentation, the context that is assigned to a queue in queue.conf is only valid before an agent has accepted the call. I'm still running Asterisk 1.0.6, which is the current version for SuSE 9.3. Maybe Asterisk 1.2.x would help? Thanks for help hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --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
[asterisk-users] Asterisk VOIP / Mikrotik
have a 10 mb ethernet connection from my ISP into ether1 on a PC - Mikrotik 2.9.23 installed. ether2 is the rest of my network behind the router. How do I prioritize packets such that VOIP calls ALWAYS get a "clean channel" through to my Asterisk server, which resides behind that router ? Things sound choppy at best at the moment. HelP! ___ --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] Asterisk VOIP / Mikrotik
Yep, using SIP for users, IAX for trunks. Can't seem to figure out how to help out the RTP streams though. Once in a while, calls seem clear, but most of the time they're choppy as anything... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Friday, July 28, 2006 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk VOIP / Mikrotik On Jul 28, 2006, at 10:55 AM, Curt Shaffer wrote: And, someone correct me if I am wrong here, you want to make sure RTP is getting quality as well. SIP is setting up, tearing down, and a few other things but RTP is where the conversation is taking place. Yes, if he is using SIP. He didn't mention that. ___ --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] Provider UNREACHABLE
Bill's right. But, it happens to me too, ALL the time, w/Teliax. I can't wait for their NYC node... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, July 12, 2006 9:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Provider UNREACHABLE It's the internet...maybe for you the path to Teliax is kinda crappy? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Wednesday, July 12, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provider UNREACHABLE Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --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] Provider UNREACHABLE
Teliax ? I'm seeing the same. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Tuesday, July 11, 2006 7:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Provider UNREACHABLE Hi All I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec apart most of the time and then sometimes for about 45 - 74 minutes I have tried a reload and sip reload but neither bring the provider back ? What else could I try and how do I prevent this Thanks in advance Barry ___ --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] Provider UNREACHABLE
teliax had a 2.5 hour outage today. I wouldn't call that short. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Paglayan Sent: Tuesday, July 11, 2006 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provider UNREACHABLE they had a short outage today, it was fixed already, dunno if related to your issue, ___ --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] Easiest (best?) linux distribution for dedicatedAsterisk box?
I'll second that. I use Ubuntu, actually installed asterisk through apt-get. Too easy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Fedyk Sent: Tuesday, June 13, 2006 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Easiest (best?) linux distribution for dedicatedAsterisk box? First, remove telnet from your vocabulary. It should only be used over serial connections these days. All other times, you should be using ssh. Second, do you want the computer to be installed and running without any major software changes for a year or more? Then use Centos or Ubuntu Dapper 6.06 or Debian Sarge 3.1. Make sure you don't install the graphics as it can affect the latency of asterisk, especially on older hardware. Third, I run asterisk on a PPro 200 at home, so your machine is beefy enough for sure. And lastly, just give it a try, you'll learn a lot just making the effort. Mike John Klimek wrote: First off, I'm sorry for sending so many messages to the list-serv. Hopefully this will be my last for a while! I was going to use my WRT54G router as a small Asterisk box, but I forgot that I had a spare eMachines computer (Intel Celeron 633 MHz, 20GB HD, 64mb RAM). Will this machine work OK for a very simple dedicated home Asterisk box? Also, what is easiest linux distribution to use and install? All I want is a simple Asterisk box that I can telnet into and have voicemail, music-on-hold (MP3), etc... ___ --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] GXP-2000 MultiPurpose Keys
good question! I'd like to know too, so keep it public please !:) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Friday, June 09, 2006 9:42 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] GXP-2000 MultiPurpose Keys Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? 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
[Asterisk-Users] Busy Signals after hangup
I've not seen an answer to this in any forum. I make a call through Asterisk, with a VOIP phone, doesn't matter which. The call gets made, I leave a voicemail, or complete the call in some manner, and the other side hangs up. I hear a busy signal on the phone on my end. If I have an extension that looks like this, after the hangup() is executed, my phone gives busy signals until I hangup and pick up to get another dial tone. exten = 199,1,Answer() exten = 199,2,Dial(SIP/100,20) exten = 199,3,Hangup why? And how to fix ? This is annoying... R ___ --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] PSTN Incoming call on real line disrupts VoIPcall over DSL circuit
what'd that fix have to do with ? Is it a frequency interference thing ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex RobarSent: Monday, May 08, 2006 4:08 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIPcall over DSL circuit I'll lean this way too. I had a DSL line from Bell Canada in Kingston, Ontario, and an incoming call on that line to the POTS phones would cause VoIP traffic to become completely unintelligble. The VoIP call would have to be re-established to fix things. A quick call to Bell had a technican out to check the lines, and put a fix in place for me. Alex Robar On 5/8/06, Jerry Jones [EMAIL PROTECTED] wrote: I would guess either the DSL itself is bad or perhaps the dsl Modem.perhaps calling Bellsouth would be helpful? Does other Internettraffic get interrupted also?On May 8, 2006, at 1:42 PM, Hadar Pedhazur wrote: I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the "real" phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection with no problems either, so the Internet connection has not been lost. I have to admit to being completely clueless as to what to even look for, so _any_ advice as to things to test for would be appreciated. As I said at the top, I can reproduce this 100% of the time, so I can easily setup any debugging environment in advance, and trigger the problem at will, etc. Thanks in advance! ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --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] Separating Asterisk SIP extensions from dialing each other.
This is coming from an * noob. :) I've got two customers, they both are replacing their phone systems with VOIP, and we need to retain both their existing dialplans. One has 5 extensions starting at 100, and the other has 10 extensions, starting at 100. Is there a way to have the same extension number twice in the same asterisk system ? They will have different incoming DIDs of course. I don't want them to be able to see / hear / feel / dial each other internally, either. They must remain completely independent. If anyone's got pointers in a Wiki or PDF somewhere, let me know. Thanks ___ --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] Polycom 501's for sale
Converted a strictly VOIP system in NYC to NEC IPK TDM system... will have 25 Polycom 501's for sale. Best offer, offlist only please. R ___ --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] Polycom 501's for sale
Customer got ripped off by a previous VOIP provider and had a REAL distaste for VOIP, even done right... Get this they had a SIP Server in San Diego, with 25 phones in NYC and another 20 in Atlanta. Average hops were 24, and over 210 ms end to end. Just poor engineering, and they didn't know better. Gabriel Afana wrote: Why the changeover back to TDM?? - Gabe - Original Message - From: Rick Smith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 23, 2006 4:39 PM Subject: [Asterisk-Users] Polycom 501's for sale Converted a strictly VOIP system in NYC to NEC IPK TDM system... will have 25 Polycom 501's for sale. Best offer, offlist only please. R ___ --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
[Asterisk-Users] LNP / DID Service - Louisianna / Virginia
I need to convert quite a few numbers in play, as Remote Call Forward Numbers and this is a sample of NPA/NXX's that we'd like to convert to VOIP right away. We are using an NEC 2000 IPS switch to do the conversion and feed to our call centers, and I will want to add DIDs from these same NPA/NXXs later... 337-774 (LA) 540-371 (VA) 540-389 (VA) 540-562 (VA) 540-953 (VA) 703-276 (VA) 703-323 (VA) 703-450 (VA) 703-502 (VA) 703-573 (VA) I have about 1,800 more locations, nationwide, that will need to be ported over time. I want free inbound calls to these numbers, 1.5 cents / min outbound from them. I'd like to take the calls via PRI in a NJ datacenter somewhere, but we could discuss SIP *trunk*ing (not handset termination) as well. Please contact me offlist if you can provide this service. Rick ___ --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] Traffic prioritization and 'class of service' for SIP
Phil; What link ? We're got a T1 from Sprint that we use for internet. During VIOP calls, if you download something, the VOIP calls break up. I found some info at Sprint for adding 'class of service', and I also have some information on configuring our Cisco routers. I've read the relevent pages on the wiki, but it seems vauge what's required and what's required by the NSP (Sprint). How have you dealt with this problem? Is this something which requires the NSP to be involved, or can this all be done on the premises side in IOS configuration(s)? Phil ___ --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] Automated Dialing / Recording ?
We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for billing and payment purposes as well as operational purposes, and I thought this would be the perfect situation ! Thanks, Rick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Automated Dialing / Recording ?
Cool... What I actually wanted to do with this is combine the * operator voice with the phone number and make a web file out of it...then let someone go down the list by browsing a website. Of course, a database app would store and create the call lists Thanks! This gives me ammo. R -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Monday, February 02, 2004 2:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Automated Dialing / Recording ? On Mon, 2004-02-02 at 13:06, Rick Smith wrote: We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for billing and payment purposes as well as operational purposes, and I thought this would be the perfect situation ! Yeah, you can do that, but what help will the recorded files be? Seems that if you are wanting to be sure a forward functioned, you would want some form of positive feedback. Anyways, you could create a context that had an absolutetimeout then dumped to a monitor app, then the timeout would hangup for you. Combine that with the sample.call files to dial out and then dump to this context and your done. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Automated Dialing / Recording ?
Awesome idea. Thanks. -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Monday, February 02, 2004 5:15 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Automated Dialing / Recording ? Maybe you could add the idea of how long is the wait time to this application and have each call go to a forwarded number, and wait around maybe playing a message for he operator on the other side to hit some DTMF key sequence that breaks the loop. Why bother having someone listen to if the call succeeded when you can get the phone user to confirm for you. It is still a simple application that you call out and connect that call to a local monitor app. If you don't get a positive acknowledgment, you could then forward the audio file and the specification off to a human to be interpreted. If your talking about 1000's of lines, this should cut down the manual labor quite a bit. On Mon, 2004-02-02 at 16:05, Rick Smith wrote: Cool... What I actually wanted to do with this is combine the * operator voice with the phone number and make a web file out of it...then let someone go down the list by browsing a website. Of course, a database app would store and create the call lists Thanks! This gives me ammo. R -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Monday, February 02, 2004 2:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Automated Dialing / Recording ? On Mon, 2004-02-02 at 13:06, Rick Smith wrote: We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for billing and payment purposes as well as operational purposes, and I thought this would be the perfect situation ! Yeah, you can do that, but what help will the recorded files be? Seems that if you are wanting to be sure a forward functioned, you would want some form of positive feedback. Anyways, you could create a context that had an absolutetimeout then dumped to a monitor app, then the timeout would hangup for you. Combine that with the sample.call files to dial out and then dump to this context and your done. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Firmware ?
I'm getting 1.0.4.30 I think it is, in new phones, but all that's on the website is 1.0.3.81 Where do you download newer versions ? And, will anyone else's firmware work on these ? This firmware seems to be flaky at best. These Budgetone phones SUCK with NAT involved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream Quality Survey.... :P
Send me all your grandstreams. I don't see anything wrong with em. (Now that I figured out how to set them up with *) :) -Original Message- From: Nick Bachmann [mailto:[EMAIL PROTECTED] Sent: Friday, December 26, 2003 4:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Grandstream Quality Survey :P Brian West wrote: Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? OK, so GrandStream phones are crap. What other phone products are there on the market that are cheap (and I DO NOT want to buy phones off eBay for a business), but work well. ATAs are out of the question because they aren't phones... and don't support all possible VoIP features. Cisco phones are all at least $200+, with maintenance, right? So what other options are there for ~$100 SIP/IAX hardphones? Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
