[asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a Nortel DMS 100 in National ISDN 2 mode. I've put some 'pri intense debug' output below. Names and numbers have been changed to protect the innocent :) Is there anybody out there using a Sangoma A10X series card on a Telus PRI in Alberta, and do you have CID NAME working? Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 043 0: 0 N(R): 039 P: 0 90 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=90 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1c 1a 9f 8b 01 00 a1 14 02 01 04 02 01 00 80 0c 41 63 75 72 65 20 48 65 61 6c 74 68] Facility (len=28, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x14, 0x02, 0x01, 0x04, 0x02, 0x01, 0x00, 0x80, 0x0c, 'Customer', 0x20, 'Health' ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 0d b1 41 63 75 72 65 20 48 65 61 6c 74 68] Display (len=13) Charset: 31 [ Customer Name ] [6c 0c 21 80 34 30 33 35 33 39 35 37 39 37] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '403814' ] [70 0c a1 31 36 30 34 32 39 38 32 37 39 34] Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '16045552794' ] -- Called g0/16045552794 pbx*CLI [ 00 01 01 58 ] You can see that it's sending both Facility IE and Display IE name information. The technician was suggesting that sending both might be the problem. If so, I have no idea how to turn off the Display IE, and I solicit suggestions :) The rest of the PRI stuff is just call setup. Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 044 P/F: 0 0 bytes of data -- ACKing all packets from 42 to (but not including) 44 -- ACKing packet 43, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 counter pbx*CLI [ 02 01 4e 58 08 02 80 04 02 18 03 a9 83 81 ] pbx*CLI Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 039 0: 0 N(R): 044 P: 0 10 bytes of data -- ACKing all packets from 43 to (but not including) 44 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 4/0x4) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Sending Receiver Ready (40) [ 02 01 01 50 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 040 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Zap/1-1 is proceeding passing it to SIP/121-082399e8 pbx*CLI [ 02 01 50 58 08 02 80 04 01 1e 02 80 88 ] pbx*CLI Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 040 0: 0 N(R): 044 P: 0 9 bytes of data -- ACKing all packets from 43 to (but not including) 44 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 4/0x4) (Terminator) Message type: ALERTING (1) [1e 02 80 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0)
Re: [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
Hi, James -- thanks for your comments. James FitzGibbon wrote: On 11/6/07, *Stephen Bosch* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. I've had no end of trouble getting CNAM out of NI-2 PRIs with Telus. We're in Ontario, but the switch configs are the same across the country I believe. Well, that's not strictly true. Configurations in BC and Alberta are different, a legacy of the pre BCTel/Telus merger days. BCTel had a lot of GTE Automatic Electric equipment in its network. It survives if it goes to a Telus customer, but not if it crosses over to Bell, Rogers, etc. Well -- here's where you can help me, because our name info is not even surviving on Telus' own network. I don't really care too much about Bell and Rogers, since Bell barely has a footprint out here and Rogers doesn't provide CNAM on its mobile network anyway (and nobody is using Rogers home phone ;) ). So -- if you had it working on Telus, what did you do? One tech claimed it was because I was sending calling name in addition to the IE, He probably meant the Display IE *and* the Facility IE. If you see my post it's what the technician I was working with suggested. Would be great if I knew a way of turning off the Display IE, if that's even possible/allowed. If it's not, then the don't send both idea is wrong. while another claimed it was just a problem when the call passes from a NI-2 circuit to NI-1 (which some of the other carriers still use). Yeah, I've confirmed that this is an issue through a number of different people. So, no real solution for you, but at least you know it's not something obvious you're doing. I've tweaked my zaptel settings back and forth and tested with Telus on the phone to no avail. In the end, we deemed the effort to not be worth it. Did you end up adding name records to the LIDB? Anyway -- again -- what did you do to get it working on Telus' network? Do you know what kind of switch you were connected to? -Stephen- ___ --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] Digium Vs sangoma Hradware
Michelle Dupuis wrote: And just for confusion, one of the guys I work with swears by Sangoma. (I have not done a lot of T1 stuff personally...so maybe as your expertise grows Sangoma becomes a better fit). Perhaps I should have voted for Heinz... as for string -- go with fishing line. -Stephen- ___ --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] Voicemail issues in 1.4.11
Jeremy Mann wrote: Asterisk isn’t playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the “The person at extension…” message, not the greetings I have recorded. Try to get rid of all of the unavail files in your directory but one. I had the same thing happen to me and realized that I had lost track of which files were valid and which weren't (IOW, I had a bunch of empty or corrupt audio files). Use the process of elimination to find a file and codec that works. You might also try doing a stack trace if increasing the verbosity doesn't help you find the problem. -Stephen- ___ --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] Is there real benefits on a SMP machine for Asterisk?
shadowym wrote: Whatever your many reasons, using that stuff for Asterisk is a waste of money but go crazy if you want! Well, all I can say is, you're clearly not dealing with my clients. They want the phones to work. Always. When they don't work, the clients get very, very angry at me. -Stephen- ___ --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] What web GUI are people happy with?
Paul Hales wrote: I use vi. Not sure if it has a web interface yet. vi is great, but our clients don't know how to use it. They also don't want to have to call us every time they want to make a change -- so for them, FreePBX it is. I make its limitations very clear to them. The other problem with vi is that there's no native version for Xbox. -Stephen- ___ --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] phone as control interface (was 99bottlesof beer)
John Faubion wrote: Does anyone know of such a device that I can use over a network? It would be a pain to run a USB cable. I am thinking of devices that are like: I think your missing the key feature of these devices, UPB/X10. Since when did Phidgets do X10? X10 is ugly, anyway. -Stephen- ___ --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] Loud pop at the end of messages causing level problems
Eric Deutsch wrote: Hi everyone, I’ve set up a little Asterisk system with a Digium TDM400P and everything works splendidly except for the messages callers leave. Every message that a caller leaves is very faint. I’ve already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? Before you tinker with the gain settings in voicemail.conf, I recommend you tweak the gain settings in zaptel.conf. -Stephen- ___ --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] DS3 Interface
Hi, Brian: Brian West wrote: Matt, I talk very openly about this issue. It was very rude of you to bring this up. This plea was total bullshit. If you want to know the whole story feel free to call me and talk about it. 918-424-9378... anyone can call me and ask me questions about it. The plea was a deal worked out between the DOJ and my attorney which was good because I signed my plea on Sept. 4th 2001. If you try to fight the DOJ you will not win. That plea was the only way to settle the issue without a trial. All I did was click edit in frontpage and alert them of anonymous publishing priv. were on their servers and they called the FBI and three days later our office was raided. This I consider mudslinging by you and wasn't very gentle man like. I'm not making excuses for anyone, but I have the following suggestion: If you treat people respectfully, without being snide or aggressive, then you will be much less likely to open yourself up to responses in kind. You throw your weight around a lot on the list. Try giving friendly, constructive input. You will be amazed at the results. Cheers, -Stephen- ___ --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] DS3 Interface
Brian West wrote: And what was the purpose of this? So that we would realize who we were talking to. :) -Stephen- ___ --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] Weatherproof Hard Phone
Philipp Kempgen wrote: Don Kelly wrote: http://www.sandman.com/autodial.html These phones look like the ones we had in Germany 20 years ago. ;-P Hey, don't knock it, Phillipp :) -- I'm as big a fan of German technology as anybody, but these phones are amazing pieces of engineering. Reliable, with excellent sound quality, and practically indestructible. There's a reason they're still in production after all these years. -Stephen- ___ --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] $70 USD bounty for simple Junghanns ISDNguard shell script
Nick Richardson wrote: Hi all, I recently purchased a Junghanns ISDNguard and to my horror I found out: - Junghanns technical support is non-existant - I can't use it without recompiling Asterisk with res_watchdog Let me know if you get any response on this bounty. Cheers, Stephen Bosch Calgary, Alberta, Canada ___ --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] Sangoma vs Digium: (was Re: ping too)
Hi, all: I think everybody is entitled to their biases, and I have to say that -- far from seeing this as a flame-war or otherwise tedious -- I think it's great that we're having this discussion and getting open and honest input from Digium staffers. We want to hear your thoughts and feelings on the issue, because the rumour mill has been going full-blast, and honesty helps public perception and keeps the speculation to a minimum. I appreciate the difficulty of Digium's position as both the keeper of the flame and manufacturer of hardware. Open source business models aren't obvious or easy. So -- now that we acknowledge that running a business is tough stuff, let's be fair and say that Sangoma faces many of the same challenges. Honestly, if Sangoma had to depend purely on Asterisk for its bread and butter, it wouldn't be around, so calling it a parasite is off the mark. The arrival of Asterisk has certainly been a good thing for Sangoma, but they are a hardware manufacturer and always have been. That's a different heritage than Digium's. Different history, different worldview, different approach. My bias is purely this: I like quality. I like stuff to work. I'll give everything a chance. If Digium has made strides in improving their product (and anecdotal evidence suggests this to be the case -- personally I haven't run any of the newer hardware yet) then that's great and I'd absolutely be willing to give it another go. Digium should be (and some of the guys there seem to be) grateful that there is this kind of competition. You can argue that competing manufacturers have benefited from the open source Asterisk, but it would be disingenous to suggest -- code contributions or not -- that the reverse is not also true. The bar got raised. Certain flaws were made obvious. And let's not forget one last thing: Asterisk's utility depends on reliable hardware. We are not in a competitive vacuum here -- if Asterisk doesn't work well because the only hardware available for it is flakey, then Asterisk, the Asterisk community, and Digium all lose. Don't miss where the competition is -- it's not the other card manufacturers. It's Cisco. It's Nortel. It's Avaya, and on some planets, 3Com/Panasonic/NEC/Toshiba ;) . This is a business *ecosystem* we're in here. If I could make a couple of suggestions to Digium, right in the open sunshine, they would be these: 1. Embrace your competitors. I realize you're already doing this to some extent -- but there's a lot of rhubarb going on about what will happen to Astricon now that Digium has bought Sokol and Associates. Make sure the other guys are still welcome to come to the dance, and let them speak, too. Everybody wants to see this thing succeed, and there's lots of room on the dance floor for everybody. 2. Communicate. I realize it's a challenge when you're busy, but I can make it simpler for you. The most important thing is responsiveness. People have to know that their input has registered, or they're going to feel ignored, they'll lose their trust and go elsewhere (this has been improving at Digium in the last 6 months, so credit to them). 3. Remember that there is a big world outside the United States. Some Asterisk users in other countries have been getting the feeling that Digium cares very little about their specific circumstances and implementation challenges. (I think Digium's figuring this out too -- the BRI card is the evidence -- but there's nothing wrong with reinforcing it. Things were not so good before). Those are my 102 cents. Again - I'm glad we're talking about this. It can only help. Cheers, -Stephen- ___ --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] extensions.conf vs. AEL
Michael Collins wrote: I just got the 2nd edition Asterisk book from O'Reilly, and was surprised to find nothing in there about AEL, except a mention of extensions.ael on page 471. This is too bad. A preliminary chapter, an intro into AEL, why it's valuable, etc. would have been very welcome. Even an appendix of a few pages with examples and references to on-line documentation would have been helpful. I don't think I want to wait for the 3rd edition. Perhaps the Asterisk Cookbook will have some AEL stuff in it... drum Our book Practical Asterisk 1.4, due out 1Q 2008, will include an AEL chapter. We actually delayed publication to get it in because we thought it was important. You can check out the work in progress at http://www.the-asterisk-book.com/unstable/ /drum Cheers, -Stephen- ___ --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] Replace full PRI with SIP/IAX trunks...YES/NO?
Jim Canfield wrote: I've been considering replacing a PRI with SIP or IAX trunks. The monthly cost difference is marginal, but it would save a bit on the hardware side and soft trunks would be easier to manage. I can't help but wonder what I would be giving up? I'd like to hear some lessons learned from those who are doing it or decided, for whatever reason, it's a bad idea. Here's what you'd be giving up: reliability. If consistent call quality and reliability is what you want, SIP or IAX on unmanaged, public networks is not for you. If you can arrange private bandwidth to a carrier's POP where you can pull telco SIP channels that get you right onto the PSTN, then great. Not many locations have this available yet. If cheap is what you want and you're not too concerned about service quality, SIP and IAX can be a good option. As it appears you're running a clinic, I would recommend against dumping your PRI. Cheers, -Stephen- ___ --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] ping
Steve Totaro wrote: must be blacklisted, i have posted like 4 messages and none are showing up. That's what I thought, too, but there's some weirdness going on with Digium's list server spam filtering. Anyway, you'll probably see this one :) -Stephen- ___ --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] PRI Setup problem
Doug Lytle wrote: Alvin Austin wrote: Thanks for all of the good suggestions. I've been able to get things working. I had been trying to use zaptel svn in order to get past error messages with compiling ztdummy.ko for the 2.6.22 kernel The newest kernel that I've been able to use with the current Wanpipe drivers is 2.6.20.1 He's using the beta wanpipe, which works with the newer kernels. Cheers, -Stephen- ___ --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] PRI Setup problem
Doug Lytle wrote: Stephen Bosch wrote: He's using the beta wanpipe, which works with the newer kernels. So am I. wanpipe-3.1.4.tgz ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.1.4.tgz Hmn -- here's what his post said: I've recompiled with the latest svn sources for zaptel, libpri, and Asterisk. Wanpipe is 3.3.0.p4. Switched the T1 cable. Same result. I don't see a 3.3.0.p4 on the wiki, but maybe it's on the ftp... -Stephen- ___ --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] PRI Setup problem
Alvin Austin wrote: I've recompiled with the latest svn sources for zaptel, libpri, and Asterisk. Wanpipe is 3.3.0.p4. Switched the T1 cable. Same result. Hmn -- when you recompiled, did you 1. clean out all the source directories? 2. remove the binaries? 3. recompile in the right order? I'm not sure using SVN is a good idea here. It should work with stable ;) Has the PRI been tested with test equipment? We should make sure there is a D channel before assuming misconfiguration. I don't think we can do even a loopback test if there is no D channel... -Stephen- ___ --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] Polycom 501 Phones Rebooting
Douglas Garstang wrote: Wow. Polycom phones are STILL doing that? I haven't been involved with Polycom phones since before January, and it was a problem back then too. Jeez... Doug -- he's using 1.6.7 firmware. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Polycom 501 Phones Rebooting
Hi, Greg: I really can't recommend upgrading to a 2.x firmware highly enough. Many people had the spontaneous reboot problems and I think they were all solved by going to current 2.x firmware. -Stephen- Gregory Boehnlein wrote: Finally, press and hold all 4 arrow keys until the phone bleeps, then capture the log files dumped to your provisioning server one last time. If the problem's not obvious from reading the logs, escalate these logs to your Polycom reseller and ask them to open a ticket with Polycom on your behalf. Of course they might recommend upgrading to 2.x ;-) Well, here is what I got. Have no idea how to read these.. 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateSetup (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateSetup, Start Timer: 1000 msecs 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateOverlap (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateOverlap, Start Timer: 3 msecs 0924095418|sip |2|177|SipCallMake 8605654321 0924095418|sip |2|177|doDnsNaptrLookup: '192.168.1.1' is an IP, bailing 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 2,Proceeding 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(3) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateOverlap) - Event (SoMediaSessEvLclNetProceeding) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateProceeding (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateProceeding, Start Timer: 6 msecs 0924095418|sip |3|177|407 challenge received 0924095418|sip |2|177|doDnsNaptrLookup: '192.168.1.1' is an IP, bailing 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 2,Proceeding 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(3) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateProceeding) - Event (SoMediaSessEvLclNetProceeding) 0924095418|so |2|177|[SoNcasC]: Receiving MsgType 0x848 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 3,NULL 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(4) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateProceeding) - Event (SoMediaSessEvLclNetRingback) 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateRingBack (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateRingBack, Start Timer: 6 msecs 0924095418|sip |2|177|SipOnEvNewCodec 101a8c0,0 PCMU/8000 18536,2232 ptime=0,dir 2 index 0 0924095418|so |2|177|[SoMediaSessC]: [0x10b42844] pps[0x10edbff0] i[0x0] d[2] p[0] pn[0] lp[2232] rip[192.168.1.1] rp[18536] dp[30333] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateRingBack) - Event (SoMediaSessEvLclNetMediaInfoInd) 0924095418|sip |2|177|SipOnEvNewCodec 101a8c0,101 telephone-event/8000 18536,2232 ptime=0,dir 2 index 0 0924095418|so |2|177|soStreamAddrSet DestIP: local RTP port=2232 dest IP=192.168.1.1 dest port=18536 (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: receive-only (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|[SoStreamC]: 1st rtp pkt rx now. 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: send-and-receive (10B359C0) 0924095418|so |2|177|[SoMediaSessC]: [0x10b42844] pps[0x10edbff0] i[0x0] d[2] p[120] pn[6] lp[2232] rip[192.168.1.1] rp[18536] dp[101] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateRingBack) - Event (SoMediaSessEvLclNetMediaInfoInd) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: receive-only (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: send-and-receive (10B359C0) 0924095418|so |2|177|[SoStreamC]: 1st rtp pkt tx now. 0924095423|sip |2|177|SipOnEvNewCodec 101a8c0,0 PCMU/8000 18536,2232 ptime=0,dir 2 index 0 0924095423|so |2|177|[SoMediaSessC]:
Re: [asterisk-users] Wondering why I can't post
Bryan M. Johns wrote: Stephen, Thanks for the heads-up on the cab ride from Phoenix to the event. I did not know it was that far. I will be coming in Wednesday morning and I may take the same route you are considering. Anybody coming in Wednesday morning that wants to split fare? The sedan service we're considering is offered by SuperShuttle; you can find them at SuperShuttle.com. They also offer a passenger van service. Cheers, -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Astricon 2007 -- does anybody need a ride?
Matt Riddell wrote: Subject: Astricon 2007 -- does anybody need a ride? Heh can't see any reason it would have been moderated! I posted it at least four times, and not one made it through. Perhaps it's a spam filter. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] DECT SIP phones
Michiel van Baak wrote: On 08:00, Fri 14 Sep 07, H?kan K?llberg wrote: On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). Hello! I would reccomend the Kirk DECT gateway. It is SIP capable and avilable for N America. We have a setup with the Skinny ( chan_sccp ) protocol and in Sweden, but I wouldn't expect any problems in NA. Our customer have used it for a while now. We use a setup like that with chan_sccp on 1.2 on one customer location. I dont know if you have NEC-Philips there in NA but they have great dect/sip setups as well. we use them in a couple of installations in medical facilities (man down, assistant call, that kindda stuff) We have Philips here, but the trouble with Philips and Siemens is that the product lines between NA and Europe are usually quite different. The European stuff complies with ISO standards and can be used legally here most of the time, but buying it in Europe and then shipping and using it here can be an issue if you run into problems. Is the DECT stuff Philips branded or NEC-Philips? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Wondering why I can't post
I've been trying to post a specific message for the last four or five days. It's on a specific topic, and I suspect the topic is the reason it is not being published to the list. Which would suggest that some kind of keyword filtering is being done, though I've rephrased the message several different ways without success. I'm sending this message to see if my new posts even make it to the list. If this one does, I'll have my answer. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Wondering why I can't post
Matt Riddell wrote: Stephen Bosch wrote: I've been trying to post a specific message for the last four or five days. It's on a specific topic, and I suspect the topic is the reason it is not being published to the list. Which would suggest that some kind of keyword filtering is being done, though I've rephrased the message several different ways without success. I'm sending this message to see if my new posts even make it to the list. If this one does, I'll have my answer. Yes this post is making it. Are you bashing someone/something? Anything in the mail likely to get someone in legal trouble? The answer is no to both questions. Here's what I'm trying to post: Subject: Astricon 2007 -- does anybody need a ride? Hi, folks: Steve Totaro and I are going to be sharing a sedan from Phoenix Sky Harbor airport to the conference hotel for the conference. We're arriving on Tuesday night. The conference hotel is 45 minutes away (assuming good traffic); the taxi fare will be a killer. As an alternative, we'll be booking an executive sedan. We'll have room for one or two more people; if we fill it to the published maximum (4 people), the cost per person will be a very reasonable 19 USD per person, not including taxes and tip. If you'll be arriving on Tuesday evening and are interested, please contact me off-list. Cheers, Stephen Bosch ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Generating an old-fashioned dialtone
Phil Reynolds wrote: On Wed, Sep 12, 2007 at 11:23:51AM -0600, Stephen Bosch wrote: It's been years since I was in the UK. I can't remember what the modern dial tone sounds like. When did it change? The first version of it appeared in parts of Sutton Coldfield in 1976, but some places still had the old tone into the 1990s. The modern one is of a slightly higher pitch than the 1976 version. Much of Europe uses a similar tone. The secondary dial tone in France (that followed use of 19 when that was the International prefix) was quite similar too. The German dialtone is a single frequency with no beat, which would sound very different from the aforementioned tone only with a higher pitch... -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] DECT SIP phones
Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. Can anyone suggest a solid alternative DECT SIP phone that is available in North America? Cheers, -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] DECT SIP phones
Dave Walker wrote: On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote: Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. S450: A recent firmware (few days old) upgrade seems to have solved the issue of being able to transfer calls. The handset still does not support 'Message Waiting Indicator, but does show missed calls. Yeah, that message transfer issue was my primary concern. I'm glad it's been corrected. I could probably live without the MWI. Where did you get yours, and where are you located? Is anybody using these phones in North America? I am using this model, the audio IMO is superb and would recommend it. Siemens phones (German phones in general ;) ) have a reputation for very clear sound. I'm also interested in the Openstage phones. Failing that, there is the Aastra 480i-CT, (which is designed for the US market), but this includes a normal deskphone. If this as good as the other Aastra products, then you can't go too far wrong. I'll be doing my first Aastra deployment shortly. Everybody I know who's used them has been very pleased. If the Aastra phone is true DECT then it should be possible to order just the handsets for it. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] DECT SIP phones
Anthony Francis wrote: Aastra now makes a full SIP DECT system with cell style seamless hand off from access point to access point. Caveat: This does not use standard wireless access points, you must purchase their access points and handsets. That's okay, it's a DECT phone. It's not supposed to use standard wireless access points. I'll look into it. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Generating an old-fashioned dialtone
Phil Reynolds wrote: Quoting Clayton Milos [EMAIL PROTECTED]: Is there a way to generate an old-fashioned dial tone with Asterisk? I'm thinking of one that sounds like: http://www.seg.co.uk/telecomm/dial tone.wav As far as I know dialtone with SIP can only be generated on the handsets. We're using Cisco 7960's with SIP firmware on them and they generate a dialtone. As far as I know I didn't mention generating it as a dialtone on a SIP phone, merely generating the tone. I can probably put it on Zap phones easily enough if I wish, but I'd need to know how to generate it first, and all I am after right now is the sound. It's been years since I was in the UK. I can't remember what the modern dial tone sounds like. When did it change? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Flash IDE
Ed W wrote: I worried a lot about the same, in the end I went for a small laptop drive for safety (it's inaudible) However, this came up on slashdot recently and if you search around the logic seems to be that: - Flash rewrites quite a few times - The good stuff has wear levelling so that most roughly speaking the whole thing should work until it suddenly all fails - Given a big enough drive with a fair bit of free space then you should find it hard to wear it out in less than quite a few years even if you are hitting it quite hard (probably multiples of this). Simply do the maths to get the rough life So basically it seems that given a large enough flash drive with decent wear levelling the lifetime should be completely ample... ...Thats the theory anyway. I feel quite bullish about the whole thing, but I think I would avoid the *really* discounted cheapo flash drives since they may not have the correct wear levelling. Decent brand names should be fine though (and you can google for details on their specs) I've had CF units fail in service, but it's true that reliability is increasing, especially as they get bigger. I would recommend going with the largest CF you can afford. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Build your own appliance concept
Jeremy P wrote: Thanks for all the good info. If you're looking for a cheaper version of the thin client you could try the t5530. It's about $300 US but it only has 64 MB Flash. A 1GB flash module is $70 US but sounds like overkill for your application. Frankly, the 70 clams is the worth time saved on stripping down your install to make it fit. Flash is so cheap nowadays that it's hard to justify the effort. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
G B wrote: Hi, I appreciate the help. I called the vendor of the card and they recommended removing all of the PCI cards on the system (including the video card), and moving the card to a new PCI slot. I did all of them together, ran the system headless, and ssh'ed in remotely. It worked! haha... This must be proof that I have purchased a real piece of @#$. Glad you said it without us having to. At least it was cheap, right? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] New Installed X100p
Steve Totaro wrote: I am the last one to pickup a manual or call tech support but yesterday I was working on a very large industrial ShopBot (It is a robot so that is cool and it does really awesome things but why I was working on it don't ask.. http://www.shopbottools.com/applications.htm ) After trying a million things, briefly looking at the manual, I called the tech support line. The guy had me check two things, change one thing and everything was joyful. Had I done that from the start, I would have saved three or four hours (I bill by the hour so it's not so bad, but I couldn't bill the full rate since conscience told me not to) Show us your Asterisk configs for the ShopBot. Can I build a dresser from payphone? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Meridian S1 to Asterisk via T1
David Gomillion wrote: On 9/7/07, *Michelle Dupuis* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: This is going into an emergency response facility...where they currently have a Nortel Option 61 (I think). They want to slowly phase into VoIP. They will need 1000 phone set capacity (assuming full migration). This can be done, and I am a proponent of Asterisk. But I don't think I would recommend it in this situation. Frankly, having a big company like Nortel to blame if/when downtime occurs would be worth the money difference to me! Would it? Having someone to blame doesn't mean you didn't have a massive outage, and also doesn't mean that the vendor you are blaming is actually going to fix the problem. Which is not to say that there aren't good commercial products that are appropriate in certain circumstances... just that people place an awful lot of faith in their service agreements, probably more than they should. What you need in a situation like the above is some engineering depth and people with lots of deployment experience. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] off-hook warning tone
Eric ManxPower Wieling wrote: The correct term for this tone is howler. I'm surprised it is not in indications.conf I recall seeing it there once, but I'm reaching into the dusty recesses of my memory right now. I noticed that all the replies to the OP assumed a SIP handset. The howler only applies to analog sets. I've made the same observation -- Asterisk is supposed to send a howler, but my phones just a get a wimpy fast busy when left off hook. Once one of our analog sets was left off hook for nearly a day before anybody noticed (How come we're not getting any calls?) How do I make this work the way it's supposed to? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] where is 1.4.12?
Jared Smith wrote: On Thu, 2007-08-30 at 08:02 -0500, Eric ManxPower Wieling wrote: As I understand it, Digium does NO formal QA testing before the free Asterisk/Zaptel/libPRI releases. Asterisk Business Edition is a different story and gets extensive QA testing. As I understand it, that's simply due to a lack of resources. At the Asterisk Developer's Conference earlier this year, the Asterisk Developers were all pretty much in agreement that more needed to be done in this area, but that it would have to be a combined effort between the Asterisk community and Digium, as Digium simply doesn't have the resources at this point to do it all itself. On IRC I have been a vocal user from hell about the QA issues of Digium open source products. I've tried to be vocal about this too. And now that I'm working for Digium, I'd be happy to try to coordinate an effort between the community and Digium to try to come up with a framework where we can all work together to make this happen. That's the spirit. I just wanted to throw in something. That a product is commercial is never an assurance that it is or will be stable. It doesn't matter if the product is from IBM (which spends a billion dollars on RD annually) or Cisco or Nortel. Commercial products break, no matter how much they cost. Most vendors do a careful job of obfuscating the instability in their own product. Depending on the depth in their technical staff, they will solve the problem quickly or slowly, or offer you some limp workaround. That is somewhat correlated with the cost and class of the product. If you believe, however, that paying for a product means that it will work reliably or as promised, you are living in a Madison Avenue-induced haze. Either way... it shouldn't be an excuse for us or Digium to accept less. Take the Linux kernel -- there is a community project with a rigorous vetting process, and I would say the Linux kernel is extremely stable. We can and should introduce a similar rigor for Asterisk. A big step in that direction is patience and focus. The creeping featurism could make way for an increased concentration on reliability. That's a development roadmap thing. Cheers, -Stephen- ___ --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] where is 1.4.12?
shadowym wrote: Then you should probably use a commercial application like the Business Edition. I've found that once I decide to go down the open source road it's a different ball game. Test with the latest and greatest release that has the features you need. If it's a fairly new release chances are it's not quite ready for prime time. Open source it not the place to be bleeding or even leading edge and expect a smooth ride. And closed source is? -Stephen- ___ --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] Canada PRI order -- anybody willing to help?
Hi: I'm doing my first PRI order for a client in Western Canada, and I have the initial setup questionnaire in front of me. It has about 25 questions on it. Some of it I understand, most of it I don't. If there are any Canadian list members out there who have ordered PRI recently and who are willing to help illuminate me, I'd be most grateful. If you've ordered from Telus, that's even better. Contact me off-list. Cheers, -Stephen- ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
Zane C.B. wrote: On Wed, 22 Aug 2007 12:37:26 -0600 Stephen Bosch [EMAIL PROTECTED] wrote: Zane C.B. wrote: 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. How is this any different in FreeBSD? Could you explain to me how else you are going to mirror an entire disk in software when your boot partition is on the disk? The raid info is done the same as on other decent system, it is stored at the in the last sector of the provider. I still don't understand what you mean by this. Something has to load the RAID engine, and if the RAID engine is sitting on root partition which is on the mirror, then it's not going to work. Are you saying that this only works on disks that do not contain the root partition? -Stephen- making a mirrored freebsd system is like this... 1: install freebsd 2: dd if=current drive of=2nd drive for mirror 3: gmirror label some name 2nd drive 4: mount 2nd drive and edit fstab to boot using /dev/gmirror/whatever 5: boot from 2nd drive 6: gmirror insert name original drive /me loves GEOM, the goddess of all disk subsystems or whatever. http://www.freebsd.org/cgi/man.cgi?query=gmirrorapropos=0sektion=0manpath=FreeBSD+6.2-RELEASEformat=html ___ --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] Speech Rec on Voicemail
Ron Joffe wrote: On Friday 24 August 2007 12:37, Ryan M. Colbert wrote: I'd be interesting in pooling resources for this. We've seen the success of Vonage's Visual Voicemail and would like to emulate a similar solution. Please define success, I have a vonage account, and the transcription is very poor at best. I was about to say -- the standards would have to be pretty low to call the Vonage Visual Voicemail a success. I will give them points for daring to try, though. -Stephen- ___ --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] which OS would be fine for asterisk
Steve Totaro wrote: But in all reality, value added features such as support and automatic updates aside, is there really a mainstream flavor of Linux that is better or worse for running Asterisk (or other apps for that matter)? I have had equal luck with all that I have played with (but not heavy load tested). I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. We've run all our servers on Gentoo with excellent results. Choose your Linux distribution for stability and ease of administration -- if it meets those requirements for you, it's a good choice. Linux is a beautiful thing. I've never had something more stable! -Stephen- ___ --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] Polycom firmware download
Hi: Doug wrote: At 13:29 8/25/2007, Al lists wrote: Thats just sad, I got SIP 2.2 from trixbox now, but still we need to have some sort of place at least for ourselves to download this stuff. Looking for boot loader now. Which version? http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html#download http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip430.html#download It's funny how every time this question gets asked, there's some smart guy (who doesn't use Polycom sets himself) who finds these links. (I'm sincerely thankful for the effort, though.) Only authorized resellers can download the current firmware from those URLs. The only guaranteed way to get the current firmware is to get it from a/your reseller. Posting the firmware packages on a third-party site is a violation of Polycom's EULA. Why do they do this? Because they want to control the sales channel. I don't agree with it, but it's how they operate. If you want a more detailed answer, ask Polycom directly, and I wish you luck. Cheers, -Stephen- ___ --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] Is it posible for an incoming to ring to Polycom and cell at the same time?
Andrew Kohlsmith wrote: On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote: dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) Will this work even if the Local is pointing to a Zap channel? As far as I know, this only works with SIP or IAX outgoing. I'm not sure where you are getting that assumption from, as I have been Dialing Zap/fooZap/bar, SIP/fooSIP/bar, IAX/fooIAX/bar and combinations of all three for the past several years. That's not what was in your example. Your example is a mix of Zap and SIP. Zap channels answer immediately, so if you do Dial() to multiple technologies, the Zap() channel will always answer first. I don't think that's what the original poster was looking for. -Stephen- ___ --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] Speech Rec on Voicemail
Ryan M. Colbert wrote: I’ve had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this not too long ago. I get requests like this all the time -- but the technology is very far from being there. -Stephen- ___ --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] Is it posible for an incoming to ring to Polycom and cell at the same time?
Anthony Francis wrote: dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) Will this work even if the Local is pointing to a Zap channel? As far as I know, this only works with SIP or IAX outgoing. -Stephen- ___ --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] phone as control interface (was 99 bottles of beer)
Jon Pounder wrote: Quoting Steve Prior [EMAIL PROTECTED]: personally my favourite still is phone in intercom mode listening at all times, if you have something to say, say it. otherwise pickup and dial for control or to talk or whatever. nothing preventing you from ignoring one of the options if you don't like it, or have a phone that supports it. Computer: close bulkheads on Deck 40! Deck 40 does not exist. Uh oh. -Stephen- ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
Gordon Henderson wrote: You do (sometimes) need the hardware RAID controller to be supported by Linux and this is a weak area. Some controllers just look like a standard drive, so they are transparent to the system, but then you need to use either the BIOS utilities to set it up in the first place, or (typically) a Windows utility, although some controllers are now being supported by Linux with user-land tools to manage and check the arrays. Most proper (ie, not fakeraid) RAID controllers support Linux now. They are practically unsellable if they do not. -Stephen- ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
Zane C.B. wrote: 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. How is this any different in FreeBSD? Could you explain to me how else you are going to mirror an entire disk in software when your boot partition is on the disk? -Stephen- ___ --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] [asterisk-biz] Skype Outage Leaves Millions Speechless
Matthew Rubenstein wrote: Imagine if the world's largest online marketplace operated the world's largest alternative (and one of the largest in general) telco and an unregulated global online banking monopoly. And the telco suddenly went down, unexplained, for hours or days. That sounds like a serious threat to global economy and security, right? If the global economy is depending on a free, unguaranteed third-party VoIP service for critical communications, it deserves to go down in flames. I don't use Skype for anything important. It's nothing more than a nice to have. A tempest in a teapot. Embarrassing for Skype and eBay? Sure! A sign of Armageddon? Hardly. If anything, this is another warning against relying on Microsoft Windows. -Stephen- ___ --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] 99 bottles of beer
SIP wrote: Russell Bryant wrote: Steve Murphy wrote: How about this one: from an extensions.conf that someone posted on the internet, I think, and I converted to AEL; I'm sorry, but I can't find the original author. (If anybody can find his post, I'd love to give him credit.) I did test this out, and it works; just put a call to the macro ( guessgame(); ) in an extension in your dialplan Nice! While we're on the subject of silly but fun dialplan bits, check out my TV remote extension. When I moved a few months ago, there was a while when I couldn't find the wireless keyboard that I usually use as my TV remote to control MythTV. So, I built dialplan so I could use a wireless phone as my remote, instead. The dialplan reads digits from the phone and sends the correct commands to a MythTV network control interface for the frontend application. I posted my tested .conf version and the untested AEL version to the MythTV wiki. The AEL version would probably be prettier with macros, now that I think of it ... http://www.mythtv.org/wiki/index.php/Controlling_MythTV_from_any_phone_using_Asterisk Wow... that's just wow. Words fail me. I'm not saying it isn't cool... just... wow. ;) It's a nerd explosion in your mouth! -Stephen- ___ --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] TDM400P FXO click sounds
Hi, Gustavo: [EMAIL PROTECTED] wrote: Hi all and thanks for every suggest about my problem, I found that my TDM400P was sharing IRQ with onboard sound device using cat /proc/interrupts, lspci -v and lspci -vb. When I disable all unnecessary hardware on my machine and test it, clicking sounds continue on the line with the same intensity; again using lspci -vb i found that: 01:00.0 VGA compatible controller: VIA Technologies, Inc. Unknown device 3230 (rev 11) (prog-if 00 [VGA]) Subsystem: Micro-Star International Co., Ltd. Unknown device 7253 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 Memory at c000 (32-bit, prefetchable) Memory at dd00 (32-bit, non-prefetchable) Capabilities: [60] Power Management version 2 Capabilities: [70] AGP version 3.0 04:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0003 Flags: bus master, medium devsel, latency 64, IRQ 11 I/O ports at be00 Memory at dfaff000 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 Now TDM card share IRQ 11 with onboard vga controller. I have a sata raid 1 level running on the box too and cat /proc/interrupts show me: 0: 23572057 0IO-APIC-edge timer 1:196 0IO-APIC-edge i8042 6: 3 0IO-APIC-edge floppy 7: 0 0IO-APIC-edge parport0 8: 0 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 66 0IO-APIC-edge ide0 209:3663990 0 IO-APIC-level eth0 217: 403070 0 IO-APIC-level libata 225: 95602389 0 IO-APIC-level wctdm NMI: 3824180 LOC: 23572106 23572083 ERR: 0 You must ignore the IRQ flag in the lspci output when your system uses IO-APIC. Your /proc/interrupts doesn't seem to show a shared IRQ... are we sure this is the real cause of the problem? -Stephen- ___ --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] Mitel 5020 IP phones
Hi: I've got a dozen Mitel 5020 IP sets and can't find out if they do SIP, or even find an administrator's manual for them. Mitel has been rather unhelpful. They only deal with partner resellers. Has anybody used these with Asterisk? -Stephen- ___ --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] Some advice
William McCloskey wrote: The stability problems we have seem to be related to asterisk crashing the apache install on the box when the PHP scripts are performing functions via asterisk. Don't know exactly how they work it all, but that's the gist of it. Are the PHP scripts connected with paging at all? -Stephen- ___ --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] Sangoma Wanpipe installation problems
Dr. Michael J. Chudobiak wrote: Rory Campbell-Lange wrote: I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm getting compilation errors when trying to install the wanpipe utilities. Sangoma says that 2.6.21/22 is not supported yet, just 2.6.20. They're working on it. In the interim the beta drivers will work. -Stephen- ___ --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] TDM400P FXO click sounds
shadowym wrote: Please explain to me how FXO tune would fix popping and clicking sounds??? If they are caused by a poorly-tuned echo canceller. -Stephen- ___ --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] Mitel IP 5020 phones
Hi, folks: I've come into some Mitel 5020 IP phones. A client has made a significant investment in them and we want to see if we can use them in a new system. Are these even SIP sets? I haven't been able to find out. Mitel's site barely covers them (I was only able to find some user guides, which are effectively useless; they say nothing about configuration and provisioning). Has anybody used these with Asterisk? Any other feedback or advice out there? -Stephen- ___ --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] Dial plan suggestions
Anthony Francis wrote: You want a key system, the fianl frontier of an asterisk implementation, and currently my holy grail. The best way to do it in an ugly way is to park the call and have a speed dial for pickup. Some phones like Aastra 55i and 57i can even have their hold button reprogrammed to blind transfer to the call parking. Isn't this what Shared Line Appearance is supposed to do? (Supported in 1.4...) -Stephen- ___ --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] Dial plan suggestions
Anthony Francis wrote: Since I dont use 1.4 then you tell me. :) This functionality is supposed to be supported in 1.4, though I've never personally tested it. When it's configured it gives the key system behaviour you describe. -Stephen- ___ --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] Pickup command
Chris Earle wrote: Not really sure about SIP exactly, but for Asterisk 1.0 versions, I know that the Pickup only works with Zaptel channels -- so to use it for any sort of IP channel, IAX for example, you have to use an addon/patch google it, 'pickup2' I think it's called works well, allows the Pickup command to grab any ZAP or IAX channel Have you used this yourself? I need something like this. (This limitation is frustrating.) -Stephen- ___ --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] Some advice
William McCloskey wrote: I need a quick bit of advice from the list. We purchased an asterisk based phone system back about 6 months ago and we are using Cisco 7940G phones (I know, not everyone's favorites). We are using the second line on the phones for paging with a auto-answer, now my question is having the system call 20 of these paging extensions, should that be enough load to cause instability in the system? Our vendor is claiming it is causing the problems we are having, and I really find that hard to believe. Can you be more specific about the stability problems? That's a bit vague -- it makes it hard to understand what's really happening. -Stephen- ___ --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] Ordering BRI From ATT
Trevor G. Hammonds wrote: I am not aware of any commercial Asterisk-compatible cards that support North American BRIs right out of the box. The best I have been able to come up with was a card sold on eBay, where the seller promises to supply a patch that needs to be applied to Asterisk (based on BRIstuff) so that it will support North American BRIs. The driver allows only one SPID per BRI, so multiple DID/MSNs are not supported. The card you're referring to is the OpenPCI card; they have a new stack that supports multiple SPIDs, which is now in beta testing. I understand that they actually have some cards deployed with US customers, too. Trevor -- are you using any BRIs at the moment? -Stephen- ___ --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] Ordering BRI From ATT
Tzafrir Cohen wrote: On Sun, Aug 12, 2007 at 12:42:10PM -0600, Stephen Bosch wrote: Trevor G. Hammonds wrote: I am not aware of any commercial Asterisk-compatible cards that support North American BRIs right out of the box. The best I have been able to come up with was a card sold on eBay, where the seller promises to supply a patch that needs to be applied to Asterisk (based on BRIstuff) so that it will support North American BRIs. The driver allows only one SPID per BRI, so multiple DID/MSNs are not supported. The card you're referring to is the OpenPCI card; Any relation between bristuff and chan_vpb that I wasn't aware of? No -- sorry, my mistake. I got the name wrong. The card is actually from PhonicEQ; there's a description of the card at quadbri.phoniceq.com. I actually don't know much about the stack. I think it's a patched libpri, actually. It's sounds interesting, though I haven't seen it personally. -Stephen- ___ --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] 20min waiting time
Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. Couldn't one change the default timeout so that Dial() will ring for 2 seconds? Or will that have all kinds of other undesirable side effects? -Stephen- ___ --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] Ordering BRI From ATT
Trevor G. Hammonds wrote: Bill, I am not aware of any commercial Asterisk-compatible cards that support North American BRIs right out of the box. The best I have been able to come up with was a card sold on eBay, where the seller promises to supply a patch that needs to be applied to Asterisk (based on BRIstuff) so that it will support North American BRIs. The driver allows only one SPID per BRI, so multiple DID/MSNs are not supported. Fortunately, PRIs are relatively cheap in California. As such, I have not yet made a concerted effort to find a card that does all that I need over BRI -- though I am really interested in having this capability. I wish the Digium BRI card had the drivers for North American ISDN. Such a shame that they went to the effort of getting FCC approval, but didn't bother to do the work to actually make it work in the US. Sangoma is willing to support NAm BRI for their new A500 BRI card, if there's enough interest. If this is something you want, you should let them know. Last I checked, there was also an informal poll on the Sangoma homepage. -Stephen- ___ --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] test the email-list OT
C F wrote: OMG, someone thought that it's for real. Wow. I don't think so. Read the sentence carefully: On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote: C F wrote: No you cant. This message is being dropped as well. Shame. Seriously though I posted a new thread right after I posted that He got the joke. -Stephen- ___ --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] FW: Can you reload only one conf file?
Mike wrote: The thing is that I make them automagically reload from outside Asterisk (by calling asterisk -rx extensions reload) Correct me if I am wrong, but can't you load and unload individual extensions from the console, or through the AMI? That's what I meant you can script this. -Stephen- ___ --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] 705 DIDs for Collingwood Ontario?
Andrew Kohlsmith wrote: On Thursday 09 August 2007 1:18:17 pm Stephen Bosch wrote: Why would anyone want a Collingwood DID? I don't answer calls from Collingwood simply because I am plain old not interested in the free vacation weekend I keep winning. :-) Are there lots of boiler rooms in Collingwood? ... Boiler rooms? (I know what they are, I just don't get the reference...) Ah... Collingwood is on Georgian Bay. Sorry. My geography isn't that good. There are some suburban areas in the Greater Toronto and Montréal areas that are popular sites for bogus outgoing call centres whose primary job is to rip off American senior citizens. Unfortunately, Asterisk was the best thing that happened to a lot of these outfits. -Stephen- ___ --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] Major Digium Card Problems
Jay R. Ashworth wrote: On Fri, Aug 10, 2007 at 07:52:30AM -0400, Steve Totaro wrote: Sure. But we were talking about installers who do it *wrong*. -- jr 'at least, *I* was' a Luckily, I was trained by a guy that had been doing telcom work for forty years. He used to be a lineman in the Philippines (no bucket trucks there, all pole climbing). He came to the US and worked in CA as a phone system installer. An absolute perfectionist. Almost to the point of being annoying but being his apprentice was more valuable than anything else I could imagine. Every business inside-wire guy I *met* in the early 80s was like that, and that was *GTE*. :-) Grounding, perfect. All mounted equipment perfectly level, all wiring and cross connects perfect. Being from LA, he taught me to always leave a loop or slack in crossconnects for earthquakes and several feet of extra coiled cable in the ceiling just in case a block needed to be moved down the road. If anything was ugly or not perfect, he would re-do the whole thing. I don't know how many 66 blocks I had to re-terminate to a 25 pair cable because it was not pretty enough. Yay! Anyways, most data guys do not understand this stuff. It would certainly make a great chapter in a future Asterisk book if the data guys took the time to read and understand it. Maybe a short segment at AstriCon or something on AsteriskTV? Indeed. I had the great fortune of being a data guy with several years of telco experience, mostly working with a top notch phone system installer. Now, I can go into any telco closet and know quite a bit about the installer's ability and work ethic. Yep. Course, some of them aren't up to it anymore, though I did see a CLEC installer do a Bell-quality job a couple weeks ago. Guys -- this is where my nostalgia comes from. -Stephen- ___ --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] Sort of OT: PBX vs CO
Alex Balashov wrote: On Fri, 10 Aug 2007, Jay R. Ashworth wrote: Short version: There's some hope Asterisk could handle the programming, but the switching fabric simply is *not* up to the task yet. And I am not sure that kind of DSP density or CPU-bound framing and transcoding is even possible. At the very least, Asterisk would have to have a vast array of rather expensive ASIC cards developed around it that would offload a great deal of this functionality; the dedicated DSP support is a good start, but nowhere near where it needs to be. I read an article about a Luftwaffe pilot who broke the sound barrier in an Me262 (the WWII jet fighter). Of course, he did it in a dive. The claim was disputed. An aeronautical engineer was quoted as saying, Even if it were true, this is a little like doing Formula 1 in a riding mower. -Stephen- ___ --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] Major Digium Card Problems
Jay R. Ashworth wrote: On Fri, Aug 10, 2007 at 09:50:49AM -0600, Stephen Bosch wrote: Now, I can go into any telco closet and know quite a bit about the installer's ability and work ethic. Yep. Course, some of them aren't up to it anymore, though I did see a CLEC installer do a Bell-quality job a couple weeks ago. Guys -- this is where my nostalgia comes from. Oh, you wanted *nostalgia*? http://www.dairiki.org/hammond/cable-lacing-howto/ http://www.tecratools.com/pages/tecalert/cable_lacing.html and a photo that I *know* the Wikipedia article got from the NANOG thread: http://www.tellurian.com/california/img_8065_std.jpg In a word: Wow. In another word: Respect. Ah... lost craftsmanship. -Stephen- ___ --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] Sort of OT: PBX vs CO
Alex Balashov wrote: On Fri, 10 Aug 2007, Gordon Henderson wrote: On Fri, 10 Aug 2007, Anthony Francis wrote: On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote: And as a CO switch, you *must* switch TDM; VoIP isn't really an option. Really? http://www.pt.com/products/prod_segway_ntwksolution.html And BT's 21cn (21st Century Network) is touted as being entirely IP, and they're rolling it out to the whole of the UK in the next few years. They've already started with a few small towns and AIUI they're working their way through exchanges as I type... My exchange is scheduled to be converted in Q1 2010. This is all good and fine. Even then, Asterisk simply won't do because of scalability limitations associated with it intrinsic programmatic characteristics as well as the hardware it runs on. It might be possible to glue something together with it and OpenSER and a media gateway control protocol like H.248 and a few of these SS7-IP appliances, but it would have all the ragtag qualities of Napoleon's army routed in Russia at the beginning of the 19th century. Or doing the Hungarian Grand Prix on a John Deere. -Stephen- ___ --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] Mitel SIP phones
Hi: Is anybody out there successfully using Mitel SIP sets with Asterisk? I hear they're not the most standards-friendly, and don't play well with non-Mitel switches. I have a pile of them and would like to see if I can use them, but not if it promises to be a hassle. -Stephen- ___ --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] Terrible clicking on T1
Gleim, Jason wrote: I thought that might be an issue too... and it was originally. When we started out, I had the Sangoma card generating the timing for the span but we could never get the d-channel to come up. Turns out that since we were connected to the PSTN, we had to let the Nortel set the timing on the span because it was receiving the timing from the CO. (Essentially the timing needed to 'flow' away from the CO) But, since we got that fixed and the span started working, I felt that timing wasn't the source of the problem. Plus, if we dump the error counters on both ends, they are not incrementing... even if the span is up for several days and we clearly have the audio problems. The slip counters, framing error, etc all stay at 0 and you would figure that if it was timing slip, those would be incrementing on at least one of the sides. Okay -- if it's not clock slip (also my first inclination): Your observation that the problem goes away after the card is disabled and re-enabled, then returns after the maintenance routine runs, is a major clue. Here's what you need to find out: -Where does the card get its configuration at start time? -What is in that configuration? -What procedures does the maintenance routine perform? In this case, you want as much detail as you can get. Talk to Nortel if necessary. As an experiment -- can you disable the maintenance routine entirely? If so, do it -- see whether the problem remains gone the following day. You want to confirm that it's that routine that is causing the change, and not something else. -Stephen- ___ --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] 705 DIDs for Collingwood Ontario?
Andrew Kohlsmith wrote: On Thursday 09 August 2007 8:15:09 am Zeeshan Zakaria wrote: Does anyone provide 705441XXX, 705444XXX or 705446XXX DIDs? This is for Collingwood area in Ontario. Why would anyone want a Collingwood DID? I don't answer calls from Collingwood simply because I am plain old not interested in the free vacation weekend I keep winning. :-) Are there lots of boiler rooms in Collingwood? -Stephen- ___ --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] Major Digium Card Problems
Jay R. Ashworth wrote: On Thu, Aug 09, 2007 at 12:19:47PM -0400, Steve Totaro wrote: I was not aware that ground wire was very expensive or difficult to ground correctly. I do not see how that adds very much to the dealer's cost. A telco-grade ground on a backboard is customarily 12-ga or larger solid copper, with no breaks at all between the backboard and either a pre-master-valve water-pipe ground, or a properly instally outdoor ground rod. Or, in some cases, structural steel. Yes correctly can be difficult to manage. Any of those can, depending on where someone was kind enough to mount your backboard, be between 1 and 6 hours of labor to do properly. I dunno about you, but I charge *extra* for that sort of work. There are also lots of crappy grounds out there. Ground is often neglected, and can be the source of really stubborn sound quality problems. People don't think much of ground (or how difficult it is to get a clean ground -- you can have a good ground and still have ground loops, especially after rainfall), which is why they rarely bother to look for a ground problem. -Stephen- ___ --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] FW: Can you reload only one conf file?
Paul wrote: Mike wrote: In the interest of making things cleaner, I'd like to know if I can just reload one single conf file. Let's say I have two files, extensions.conf which includes small_file.conf. I only want small_file.conf reloaded, not the main file. Is this at all possible? Mike Not possible without modifying the asterisk source code. Yes, it is possible. asterisk*CLI help reload Usage: reload [module ...] Reloads configuration files for all listed modules which support reloading, or for all supported modules if none are listed. If you only want to reload configs for a specific module, just reload that module. Example: asterisk*CLI reload chan_sip.so -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI -Stephen- ___ --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] FW: Can you reload only one conf file?
Mike wrote: In the interest of making things cleaner, I'd like to know if I can just reload one single conf file. Let's say I have two files, extensions.conf which includes small_file.conf. I only want small_file.conf reloaded, not the main file. Is this at all possible? I'm not quite sure why you would want to selectively reload _parts_ of your dial plan, since even a big dialplan takes fractions of a second to load... Failing that: reload pbx_config.so just reloads extensions.conf. A wrapper script might be another way of doing it, but that's beyond my knowledge. -Stephen- ___ --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] Paging Application - Polycom 601
Bill Andersen wrote: Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a Page. From what I understand, the Page is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. I would be curious to see how you set up the phones to accept paging, just to make sure there isn't something iffy with your phone configuration. The problem I am having is about 1 out of 25 pages will crash the Polycom 601 (receptionist) and the phone will reboot. Is the 601 calling the page, or receiving a page from another phone? This leaves all the extensions in the conference room and each party must hit end call on their phone to get out of the conference. However, the receptionist can't do that because that phone restarts. Once it has rebooted, it does not show to be connected to the conference room. However, I feel like it is still in the conference - with no way out. You feel like it? Do you know for sure? If the phone does not show an active call, it's not connected to anything. I don't see how it would be in a conference after a reboot. Your problems below are probably caused by something else. The spontaneous reboot is telling. After one of these crashes, the 601 phone will start having one way audio (can't hear caller), various other weirdness (side car status wrong) and the only way to completely correct the problems are to restart asterisk - which I assume kills the rogue page application. The 601s with sidecars have been problematic. What Polycom firmware are you using? 1) Has anyone ever seen this problem? Other users have reported problems with 601s crashing. Check your firmware. AFAIK, the current firmware is 2.1.3. 2) Is there a way from the CLI to show and kill a page? 'show channels' will show you active calls (in 1.2; in 1.4, use 'core show channels') 'meetme kick' lets you kick channels/users from a conference. Still, I don't think that's what's happening here. -Stephen- ___ --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] les.net losing DID's
Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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] VoicePulse Connect
Wes Baehr wrote: I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren’t on my end). However, I don’t use IAX anymore, so I am not aware of any current issues. This is interesting information -- I've had similar problems with IAX trunks on totally different carriers. Example: Callers do not hear the remote ringing, or only some of the rings, or don't hear the beep tone for voice mail. IAX is easier if you're behind a firewall :( -Stephen- ___ --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] MoH mysteriously stopped working
Jay Moore wrote: Peder, Unfortunately, this did not work. Any other thoughts? My dumb questions will follow. Jay Moore wrote: Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI moh reload voip*CLI 1 class reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:422 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' I think it is trying to play mp3 files. Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:504 monmp3thread: Unable to spawn mp3player musiconhold.conf: - [default] mode = quietmp3 Is this mode appropriate when you're using gsm audio files? directory = /var/lib/asterisk/mohmp3 random = yes I have had .gsm (and only .gsm) files in that directory since day one, and it's always played them just fine. The .gsm files are still in that directory, and transferring them to my computer and playing them works just fine. I have autoload set in modules.conf, and I can't figure out why my music on hold suddenly stopped working. Obviously, something has changed. Are you absolutely sure that: - you had only gsm files in that directory - your system isn't configured to use mp3 MOH files? There's been a Windows virus going around again that deletes all the .mp3 files it finds. Is this system running Samba? What happens when you put an Asterisk ready mp3 file in that directory and restart Asterisk? -Stephen- ___ --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] Sangoma BRI card -- National ISDN/North America support (Having problems with analog disconnect supervision?)
Hi, folks! Sangoma has an informal user survey up on their home page (at http://www.sangoma.com) asking if people would use their A500 card for North American BRI, if it were supported. I encourage anyone with an interest in voice BRI in North America to vote; this information will be used for deciding whether to make the investment in developing a National ISDN driver layer for it. Benefits to users of BRI include the vastly better call quality and call control signalling, without having to pay hundreds of dollars a month for a fractional T1. (Say goodbye to analog lines left open and the resulting busy signals!) For anyone who is curious: Sangoma is writing their own driver layer for this card, not using bristuff or woomera. In other words, if they decide to go ahead with it, this would be a new contribution to the marketplace, with the bonus of having Sangoma's solid drivers and excellent support behind it. Just having decent, affordable hardware will remove a big roadblock to going digital for sites with fewer than 10 channels. Full disclosure: I would like to see this happen, as I've got plans to move our own offices and some client offices to BRI. I encourage you to vote, as well as offer Sangoma your personal feedback, if you can spare the time. Cheers, -Stephen- ___ --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] Teliax Quality of Service
Douglas Garstang wrote: Yes, all the equipment was located at the same physical location. In hindsight, we could have multi-homed our collocations. Why can't service providers multi home their edge systems to accept incoming calls from two physical locations? If a service provider did this, they would have two completely independent facilities, potentially thousands of miles apart, connected to different upstream providers. I can't think of anything short of nuclear war that would destroy their ability to accept calls. If they did least cost routing, it wouldn't even matter if their providers failed. China gets hit by a meteor and NO provider can deliver calls to China? Fine... at least you can still call everywhere else. Because all this extra expense still doesn't protect you from last mile failures. If the Internet were perfectly distributed and each node had connections to half a dozen other nodes, then maybe this would make sense, but a large amount of traffic still goes through single points of failure, even on the big Internet (Case in point -- traffic from Seattle to New York still goes through a single path south through California, across the southwest, then up and over via Illinois). When that path breaks (and it has) absolutely *everything* breaks. It's no different in the PSTN. Some people are in the fortunate position of being in areas that can be multiple-provisioned, but millions of people live at the network edge where that's of questionable value. I admire your CLECs redundancy; the security you perceive it gave you, however, is illusory. -Stephen- ___ --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] Teliax Quality of Service
Brian Capouch wrote: Stephen Bosch wrote: PSTN service still sets the standard. With infrastructure paid for under a gracious guaranteed-profit monopoly by ratepayers, In a regulated marketplace with legislated minimum service levels. In Canada, most of the phone systems were government-owned. It was a good system, at least from the point of view of reliability. I don't miss the surly (and often slow) service, but it's arguable whether today's service -- in which everyone smiles nice and *pretends* to serve you while ignoring you completely -- is any better. At least the bloody stuff worked. Communications infrastructure is a strategic, national asset, and only really useful if it goes everywhere, even to the unprofitable pockets like Podunk Corners, North Dakota. People forget this. In a totally free marketplace, Podunk Corners waits years for service and gets tin cans and string when it finally arrives. now being used as a weapon to stifle competition from VoIP, cable, and other emerging technologies. Is it? Maybe -- in some circumstances. The history of this makes for some pretty distorted economics, if you ask me. If you want an example of what happens when you don't have regulation to build infrastructure, look at Africa. All wireless, all horribly oversubscribed, and correspondingly unreliable. That's how you pay for expensive equipment in a free market. -Stephen- ___ --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] Teliax Quality of Service
Douglas Garstang wrote: So you've never gotten a dropped call or dead air on a PSTN call? Put it in a little perspective. I can count on one hand the number of outages of this kind that I've had on PSTN in my lifetime. Your mileage may vary. -Stephen- ___ --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] Teliax Quality of Service
Mark Coccimiglio wrote: Single point of failure should NEVER completely disable your company. Yes outages happen and backhoe's cut fibre all the time. From within this stuff can make one's life rather difficult, but from the outside it should be almost unnoticed. When was the last time you noticed an outage at Google, Microsoft or the DoD? Do you think they don't happen? Of course not -- but how many hundreds of millions have been invested in their infrastructure? -Stephen- ___ --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] Teliax Quality of Service
Douglas Garstang wrote: Stephen Bosch wrote: In Canada, most of the phone systems were government-owned. It was a good system, at least from the point of view of reliability. I don't miss the surly (and often slow) service, but it's arguable whether today's service -- in which everyone smiles nice and *pretends* to serve you while ignoring you completely -- is any better. At least the bloody stuff worked. Communications infrastructure is a strategic, national asset, and only really useful if it goes everywhere, even to the unprofitable pockets like Podunk Corners, North Dakota. People forget this. In a totally free marketplace, Podunk Corners waits years for service and gets tin cans and string when it finally arrives. I disagree. There is more competition in smaller towns and rural areas. It isn't cost effective for the bigger carriers to move in, so the small ones do. They get state/federal subsidies. That's exactly my point. I said: In a totally free marketplace, Podunk Corners waits years for service... A subsidized marketplace is not a free marketplace. Whether you do it with regulation and sanctioned monopoly or with subsidy, that is still a market intervention. I can't see any other way that service to sparsely populated areas would be financially viable. I'll bet you there's more ISP's, and CLEC's per square inch in Montana than there is in the bay area. Oh, I believe you. But ultimately -- what do you mean by competition? Who owns the cable plant? I have a hard time believing that there are any areas in Montana with redundant last-mile infrastructure. -Stephen- ___ --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] Teliax Quality of Service
Eric ManxPower Wieling wrote: Douglas Garstang wrote: Let's assume for a moment that it's impossible. That does not mean adding additional servers and additional networking equipment does not add value, or is a worthless endeavour. I agree with that. At least two people that I know run ITSPs. Each time they have an outage (which is not very often) they DO learn from the experience and work to avoid a future outage cause by the same issue. You would be surprised at how many little things can cause an outage. My own experience is that increasing failover redundancy, which adds correspondingly increasing complexity, also increases the odds of an outage. It is very rare that failover redundancy works as intended during an actual failover, no matter how many times you simulate it. I would rather have a simple network design where the cause of failure, when it happens, is obvious and quickly corrected. For example, I would rather have replacement parts on the shelf and be able to slap them in quickly than be running hot standbys and paying for the electricity, and then have the thing break anyway when there's a failure. -Stephen- ___ --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] Sangoma PRI
Kevin P. Fleming wrote: Have you looked at the Sangoma cards and the Digium cards? Did you notice that *both* of them are based on large Xilinx FPGA parts? They both use an 'FPGA architecture', at least for the PCI interface and TDM/data buffering (both cards use dedicated T1/E1/J1 framer chips, because it would be silly to not do so G). No, I hadn't taken a close look at both cards; Thanks for correcting me. What's noticeable about the Sangoma cards is that, when you look across the product line, the cards have the same basic frame, and the modular design is really elegant. I'm just admiring fine design. -Stephen- ___ --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] Teliax Quality of Service
Douglas Garstang wrote: This might work for a web service, but people have a zero tolerance for no phone service. They expect to be able to pick up their handset, and get a functional dialtone immediately. Adding additional servers, additional network components, and some smarts into your design saves being woken at 3am when a server fails. Frankly, if up-time is that important, then voice-over-IP is out of the question anyway, and we don't even need to be talking about network redundancy. PSTN service still sets the standard. -Stephen- ___ --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] Royalty for On Hold Music ?
John Novack wrote: The fact that ASCAP goes on campaigns doesn't make it any less absurd (or, for that matter, any more likely that the average business is going to be taken to task); the reality is that thousands upon thousands of interconnects install PBX systems with radio ports on them that are plugged into cheap transistor radios bought at Wal-Mart and similar places, and nobody -- not the client, nor the interconnect -- has any clue about any royalty obligations that entails. People do it, think nothing of it (not least because the PBX vendors promote it as a feature!) and I think neither ASCAP nor any other royalty agency has the necessary resources to make even a dent in this kind of use. Simply put - tell it to the judge. As soon as I see one, I'd be happy to. Drivers speed , change lanes, cut others off every day and MOSTLY get away with it. Doesn't make it legal, does it? The difference between that and the piped-in radio is that drivers who speed, change lanes and cut others off *know* they are breaking the law, and most people who pipe The Fuzz 104 into their waiting rooms neither know they are breaking the law, nor do they much care. They can switch to NPR if they get a letter. Seriously -- this is totally unenforceable, and most reasonable people would take a legal threat to stop listening to the radio (which is how they're going to see it) as ridiculous and insulting, even if they *do* end up complying. Not any different than stealing software is it? I happen to think that listening to commercial radio broadcast over public airwaves, whether it's over the speaker in the ceiling or the radio on my porch, is a whole lot different from stealing software, yes. It's one thing if you're Dell or Microsoft and you are using music for your call centre, and another if you're the neighbourhood dental practice. In the eyes of the law, it makes NO difference. Lots of things are ugly in the eyes of the law. That doesn't change how people actually behave. Only real consequences do. I'm talking about what is happening on the street here, not the world as you prefer to see it. I have no trouble seeing the dollar signs in the eyes of the legal barracudas on the payroll of the various licencing agencies; that doesn't make their enforcement right, reasonable, or actually happen, for that matter. There are practical limitations on how many Mom and Pop operations they can go after. Do it until you are caught, you say? Hey -- *I'm* not doing it :) I'm just looking around at the thousands of people around me who are. The music business has a horrible public perception problem, and also an enforcement problem. Chasing after people who are piping commercial radio into their premises only alienates more of the general public, the very people they are trying to get to buy their product. I'm merely relaying the reaction of the average independent business person to such a request: You want me to do *what*? Come *on*. I'd be interested in getting in touch with any small businesses which have been given a cease and desist letter or demand for payment because they piped radio into their phone systems. Not only their phone systems but their waiting rooms Next time you go into an office or store and you see the yellow ASCAP label on the door, you know they probably have gotten a letter. I have never, ever seen such a label on the door of any professional office. Feel free to introduce me to someone who has one (and I'm not kidding.) MANY interconnects now have discovered they can make extra by selling a message on hold system that not only hawks the wares of the firm but escapes the clutches of ASCAP. Introduce me to some. I'm always keen to learn. You remind me of a friend who enjoys a good argument with a tree stump. I only argue with stumps that talk. *You* remind me of the guy on the freeway who calls the highway patrol because somebody cut someone else off. I felt compelled to speak up because I see a certain constituency that snaps to salute when big money waves an attorney's letter in their faces. There are lots of laws on the books that nobody pays heed to anymore, like town by-laws which say the mayor has to give a guy he's just kicked out of town a horse and a week's rations. Laws are written by people for people (more often, by people to serve the interests of certain other people) and for specific contexts and circumstances. They serve a purpose. They are not stone slabs that Moses brought down from the mountain. That's the reason why community standards matter in the enforcement of the law. As they say in the military: the map is not the territory. Again, please introduce me to someone who's been threatened or served because they were piping radio somewhere (even better, someone who has lost a court action because of it). I would like to be educated. -Stephen- ___ --Bandwidth and Colocation
Re: [asterisk-users] Sangoma PRI
Steve Totaro wrote: Note to Digium I wish I could upgrade my wct4xxp drivers locally. I still have the v1 firmware on my card. It is kind of hard (next to impossible) to pull it from a production machine and ship it to Digium. That might take a week if all goes well. The only way this will ever happen is if Digium completely redesigns the card, which is a long way of saying that you will buy a new card before you have that request filled. This is one of the great things about the Sangoma hardware -- it was designed to be fully field upgradeable (they use an FPGA architecture). The design approach is worth emulating. -Stephen- ___ --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] Configuring Sangoma A101D with Asterisk 1.2.18 zaptel-1.2.17.1
Deepak Naidu wrote: It would help to know exactly what Dell Poweredge you were considering. They do vary. I have Dell Power Edge 850 Also how do I enable DTMF hardware detection. There are no drivers which support it. I have the lastest Beta drivers installed, they seem to show yes in the logs, but the hardware DTMF didnt work, so I wrote a mail, to the developer of the drivers he said they are still working in the lab probably have one within a week. You should try relaxdtmf=yes in zapata.conf first. -Stephen- ___ --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] polycom custom ring tones (slightly OT)
Doug wrote: At 21:59 7/29/2007, Paul Hales wrote: I even got a Polycom here saying I'll be back which was funny for about an hour, then not funny at all. PaulH Kewwl! How do you get the .wav files into the Polycom? If it's not obvious, I'd be interested in this information too. Most people seem to think you can't change the ringtones on the Polycom sets. -Stephen- ___ --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] Royalty for On Hold Music ?
Steve Kennedy wrote: On Tue, Jul 31, 2007 at 05:22:20PM -0400, Jon Pounder wrote: Quoting John Millican [EMAIL PROTECTED]: there are plenty of radio stations with internet feeds of their audio, piping that in would not change any coverage area since anyone with internet could listen anywhere already, you're only providing that to the listener through a phone handset instead of a computer speaker, which amounts to just another audio device controlled by an internet connected computer. No it's not, you're rebroadcasting and that would incur a difference license (if legal at all). What if the radio is on in the background when I make a call ? is that rebroadcasting ? kind of gets blurry on the definitions there. That's not as you're listening to it and not trying to rebroadcast. Well, this is approaching the absurd. Do you know how many Meridian systems have radios plugged into them for on-hold background sound? Nobody pays royalties on those. There are the rules and then there are the practical realities. -Stephen- ___ --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] Teliax Quality of Service
Douglas Garstang wrote: I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Don't be confused. The answer to most of your questions is no. Barriers to entry are too small for ITSPs, and there are lots of basement operations masquerading as big carriers. -Stephen- ___ --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] Royalty for On Hold Music ?
John Novack wrote: Stephen Bosch wrote: Well, this is approaching the absurd. Do you know how many Meridian systems have radios plugged into them for on-hold background sound? Nobody pays royalties on those. IF they are discovered by ASCAP and receive a letter demanding payment they will. Not absurd at all. Simply because many do it in ignorance doesn't make it legal ASCAP goes on campaigns on a regular basis. Home residential users are probably safe though not legal. Business users have a greater visibility though There are all sorts of royalty free music sources available. No excuse not to use it. Or simply pay the yearly fee to ASCAP ( in the US ) The fact that ASCAP goes on campaigns doesn't make it any less absurd (or, for that matter, any more likely that the average business is going to be taken to task); the reality is that thousands upon thousands of interconnects install PBX systems with radio ports on them that are plugged into cheap transistor radios bought at Wal-Mart and similar places, and nobody -- not the client, nor the interconnect -- has any clue about any royalty obligations that entails. People do it, think nothing of it (not least because the PBX vendors promote it as a feature!) and I think neither ASCAP nor any other royalty agency has the necessary resources to make even a dent in this kind of use. It's one thing if you're Dell or Microsoft and you are using music for your call centre, and another if you're the neighbourhood dental practice. I'd be interested in getting in touch with any small businesses which have been given a cease and desist letter or demand for payment because they piped radio into their phone systems. -Stephen- ___ --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] Teliax Quality of Service
SIP wrote: There are also lots of big carriers masquerading as big carriers. ;) *lol* If the ONLY people who could get into the business were the ones who could, before offering any services to customers, afford to build out multiple edge systems for accepting incoming calls, each with multiple interfaces connected to multiple subnets via multiple switches using multiple upstream providers, you would have ONE single choice for an ITSP. And ATT doesn't have that amount of redundancy in their network. Working in the carrier networking business, I can assure you that we've NEVER run across a SINGLE carrier network (not from the largest to the smallest) that has redundancy in ALL aspects (or even MOST aspects) of its network. This is why there are uptime policies that allow a percentage of outages to occur. Triple 9 uptime (Exceedingly rare, but a purported goal -- 99.999%) still allows 15 full hours of downtime a year. And that rarely includes the occasional lost packet or latency. In other words, you can blame the marketing departments in various big carriers for creating these unrealistic expectations in the marketplace :) -Stephen- ___ --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
