Re: [asterisk-users] Phone with public address functionality
It appears that this has been fixed on the SPA-942 but not the 941. Hopefully that will come soon. Thanks. Doug [EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Tuesday, December 04, 2007 12:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phone with public address functionality I think the newer version of the firmware fixes this problem. Paul Hales AsteriskIT On Tue, 2007-12-04 at 00:13 -0400, Doug Meredith wrote: I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. ___ --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] Phone with public address functionality
I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. Thanks for any help you can provide. Doug ___ --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 sip peer/user matching methods forauthentication backwards?
Hi, I too have found this matching to be frustrating. I would like it to behave as you describe. Doug -- Doug Meredith 506-854-7997 ext. 801 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Thursday, January 04, 2007 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk sip peer/user matching methods forauthentication backwards? Take an example where there is two sip users defined in sip.conf as follows; [peer1] Host=192.168.1.1 ... [peer2] Host=dynamic Secret=password ... [Peer3] Config not relevant ... The intention is to accept calls from peer1 without authentication (ip address authentication only), but require authentication from peer2 If by chance a SIP invite comes From [EMAIL PROTECTED] (where the name peer2 on the calling server coincidentally matches a defined sip user on the called asterisk server) To [EMAIL PROTECTED], Asterisk will attempt to authenticate the caller peer2 rather than accepting the call based on the fact that it came from a trusted Ip address defined for peer1. Since peer1 is trusted it is not sending credentials and will have its invite rejected with a 407 proxy authentication required when it fails to authenticate as peer2. This logic seems backwards to me, the IP address should be matched first, and if there is no statically defined user with that IP address the username should be matched next. This would insure that all calls from the trusted IP address are accepted regardless of whether there is coincidently a SIP user with a matching name defined on the target asterisk server. So rather than looking for a match in this order; 1. name portion of From URI in the invite (host in the URI [EMAIL PROTECTED]). 2. ip address statically assigne for a user it should look in this order; 1. statically defined sip user ip addresses 2. name portion of the From URI Can anyone shed any light on this, or suggest a workaround so 407's are not sent if the invite from header happens to have the same name portion of the URI as a defined sip user on the target asterisk server ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: My Phone Review- Large Scale Corp Deployment.
Eddie Johnson Jr [EMAIL PROTECTED] wrote: Did you test Snom or Sipura hard ip phones? I was considering Budgetone for an office of 10 users. After reading your testimonial I will have to re-think my selection. I don't know if you have used a BT but this isn't a phone that I would put in my garage much less my office. Poor sound quality, lousy display and a horrible interface. If you want something cheap to prove that VoIP actually works, then sure. If you plan to use it to talk to someone, then look elsewhere. Some other quick notes: Aastra 480i works pretty well. Sipura SPA-841 isn't much of a phone. Linksys SPA-921 941 work pretty well but suffer from occasional lockups and spontaneous reboots (latest firmware). Linksys WIP-300 is crap. Doug -- Doug Meredith 506-854-7997 ext. 801 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BRI Newbie - What Hardware, PCI, in the US?
Brent Torrenga [EMAIL PROTECTED] wrote: We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am very interested in this too. My main motivation is to get the improved signaling. You got a number of answers, but it wasn't clear to me which of them were actually in the US. My vague recollection was that the Junghanns cards weren't supported in the US. Anyone know if this is still the case? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SPA-941 stutter tone
Kerry Garrison [EMAIL PROTECTED] wrote: I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? I can't help, but I do understand your pain. I tried to turn this off with the SPA-2000 with no luck. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SPA-941 stutter tone
Jock W. Shirey [EMAIL PROTECTED] wrote: I just double checked my SPA-841. You can change the dial tone in the Web config on the Regional page. I just copied the Dial Tone: to the MWI Dial Tone field and it didnt stutter after that. I'm not sure if its the same with the 941, but i've heard the phone configs are similar. Hey, I never thought of that. One thing to check: I always assumed (but never checked) that you couldn't dial until the stutter stopped, and it gave you the normal dial tone. Is this true? If so, it will be very confusing when you try to dial when you have voice mail. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: I need suggestions for on equipment
Martin Joseph [EMAIL PROTECTED] wrote: On Nov 22, 2005, at 11:08 AM, Doug Meredith wrote: hugolivude [EMAIL PROTECTED] wrote: You need to be careful when buying the Linksys because version 5.0 saw a move from Linux, which runs Sveasoft's Talisman firmware, to VxWorks, which does not. Why would I care what OS an embedded device uses? Is there a difference in the externally observable behavior? Did you read the text you quoted? Ah. I did read it but I didn't understand it. When I read it I took it to mean that the Talisman firmware was part of the Linksys Linux offering. From other posts I now understand that it is in fact a replacement firmware. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Aastra 1.3 firmware
Lee Archer [EMAIL PROTECTED] wrote: As always right after asking it works I guess you should have asked sooner. :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Bad Lines - What can the phone company do?
Andrew Latham [EMAIL PROTECTED] wrote: Claim that emergancy health equipment does not function, that will put them in action. Better yet tell them that 911 is not captured! I'm going to have to remember that one! Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: I need suggestions for on equipment
hugolivude [EMAIL PROTECTED] wrote: You need to be careful when buying the Linksys because version 5.0 saw a move from Linux, which runs Sveasoft's Talisman firmware, to VxWorks, which does not. Why would I care what OS an embedded device uses? Is there a difference in the externally observable behavior? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can anyone explain reason for this echo
Andrew Kohlsmith [EMAIL PROTECTED] wrote: It doesn't really matter whether you buy it (my explanation) or not -- if your specific echo is greater than what the software and/or hardware are designed to handle, it will work poorly. It's called a misapplication of the technology. Two products are both intended to eliminate echo, and product A, due to it's design, can't eliminate some of the echos that product B can. It seems quite fair to say that B is a better product than A. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can anyone explain reason for this echo
Andrew Kohlsmith [EMAIL PROTECTED] wrote: I took exception to your painting the Digium hardware echo can module and the software echo cans in zaptel as trash, as they work very well for many people. They clearly aren't sufficient for your specific needs, and thus the Orion Telecom echo canceller is better -- and I stress this -- for you. That wasn't me. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ip phone
stevanus [EMAIL PROTECTED] wrote: Maybe grandstream budgetone 100 series will fulfill your requirement. It's very good for such a cheap sub-50 phone. We have two of these and they are the VoIP equivalent of a $10 K-Mart phone. I won't even use them in my house, much less the office. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk 1.2 Released!
Asterisk Development Team [EMAIL PROTECTED] wrote: We are proud to announce that Asterisk 1.2.0 has been released! That is great. I might get a chance to try it out later today. (Note: for a short time, a tarball of Asterisk 1.2.0 was present on the FTP servers with a build problem related to the chan_modem drivers; this has been corrected, and if you downloaded the new version before receiving this announcement, please re-download to ensure you have the proper version.) Just a configuration management note. The normal (and safe) practice would be to make the second copy 1.2.1. Once 1.2.0 has been released, you can't change it. It is done. Calling the second copy 1.2.1 would have eliminated any possibility of confusion. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can anyone explain reason for this echo
Eric Bishop [EMAIL PROTECTED] wrote: If I call our Asterisk box via Disa and then place a call to one of the problem analogue numbers (native Zap bridge) I don't get any echo. So the echo seems to occur only when using a SIP handset and making a call to an analogue number. The echo is probably always there. You only notice it with the SIP phone because of the additional latency that this introduces. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: call levels
Mariano Gonzalez [EMAIL PROTECTED] wrote: I need that the first phone makes calls to local numbers only and the second phone make calls to all numbers. Research extensions.conf and/or dial plan. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Message waiting notification
Sixto Diaz [EMAIL PROTECTED] wrote: i want to ring the phone user or change the tone is this posible with mailbox= ? These would be settings in your UA, not in Asterisk. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Why n s priority in CVS but not in release?
Kevin P. Fleming [EMAIL PROTECTED] wrote: actual release that will contain these features will be 1.2.0, scheduled for release within the next two weeks. Oh my God! A Date! :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP Disconnect Supervision
Steve Blair [EMAIL PROTECTED] wrote: I have a case where a call from the PSTN to our SER proxy goes unanswered. As a result it is relayed to our Asterisk server for voicemail. However before the greeting plays the caller hangs up. This results in an empty message being created and emailed or the mwi gets activated. I saw a few posts about Disconnect Supervision and Disconnect Supervision with inbound SIP connections but I did not see any resolution to the SIP question. Does this sound like a SIP Disconnect Supervision issue? If not what seems to be the issue? Also does anyone have any suggestions on how to stop these messages from being created? I think you may be misunderstanding this a bit. Disconnect supervision is a PSTN issue not a SIP issue. Presumably your problem occurs because the caller hangs up and the telco doesn't signal this to you in a timely fashion. Therefore your device doesn't go on hook and it doesn't tear-down the SIP session. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: when is 1.2 being released?
Olle E. Johansson [EMAIL PROTECTED] wrote: Adam Moffett wrote: does anyone know when 1.2 will no longer be beta? The quick answer is: When it's ready for release. Open Source software doesn't really follow a set agenda. I don't think that is an accurate statement. It is certainly true of Asterisk, but look for example at Eclipse. They produce an awesome plan up front with concrete dates. Different projects work in different ways. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream GXP-2000
Erick Baum [EMAIL PROTECTED] wrote: We're having a rather serious echo problem using the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it every once in a while on outgoing calls through the PRI. It's not the speakerphone echo problem, we're running the Crazy idea, but are the two end-points on the SIP-SIP calls nearby? Is it possible that the mic on phone A is actually picking up party B? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Please recommend a phone
Jesus Mogollon [EMAIL PROTECTED] wrote: I'm in need of a phone that would blink a led to let the callee know that there is an incoming call. The GXP-2000 does this but I want an alternative to Grandstream. Any help is appreciated. The Aastra 480i does this. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Why Asterisk documentation is so poor...
Olle E. Johansson [EMAIL PROTECTED] wrote: Sergey, I am sorry if you took our comments that badly. I proposed a worthing and you did not accept that and refused to update according to our suggestions. Tilghman therefor decided to close the bug. I suggest you try again, re-open the bug, fix the problem and continue to add more documentation. We do need more documentation! It has to be correct though, and that's why we are giving feedback. Olle, I believe I understand and share Sergey's confusion. Maybe it is something we just don't understand about how Asterisk development works. If he has made a useful contribution with the exception of one sentence, why don't you just change that sentence and apply it? Will you only accept suggestions in the form of directly appliable patches? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: PRI echo issues: solvable?
Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 18 October 2005 12:18, Doug Meredith wrote: Andrew Kohlsmith [EMAIL PROTECTED] wrote: I've never seen that, it's always when we call out. Certain numbers will always trigger it. 888-737-4787 (IPC Resistors, it dumps into an IVR so it's safe to call) is one such number, but we have local numbers that hit other I just tried this number, and it was answered by a person. It's IVR most of time time. :-) Did you hear echo? No, no echo. But I have an analog PSTN connection, not PRI. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: PRI echo issues: solvable?
Andrew Kohlsmith [EMAIL PROTECTED] wrote: I've never seen that, it's always when we call out. Certain numbers will always trigger it. 888-737-4787 (IPC Resistors, it dumps into an IVR so it's safe to call) is one such number, but we have local numbers that hit other I just tried this number, and it was answered by a person. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Canadian Association of VoIP Providers
John Lange [EMAIL PROTECTED] wrote: My apologies for the cross-posting. If you think you should apologize for it, don't do it. If you think it is okay to do it, don't apologize. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sip register incoming call contexts?
Steve Gladden [EMAIL PROTECTED] wrote: Am I still doing something wrong here? I don't have any advice to offer, but I can sympathize. This is a poorly documented area of Asterisk. I think it is quite a poor design from a SIP point of view, although it may make some sense for a monolithic PBX. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: parameters documentation
[EMAIL PROTECTED] wrote: Anyway, if you try to search insecure=very on www.voip-info.org, you find 742 links , a bit more for me. (I just want to know what it means) I think the search is broken there. Just go in under Asterisk and look for where the configuration files are documented. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: www.openpbx.org
Matt Riddell [EMAIL PROTECTED] wrote: I am astounded by the total lack of integrity people have displayed here. Isn't that a bit over the top? If you have a license that permits you to do something, and then you do it, what is the issue? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: www.openpbx.org
harry gaillac [EMAIL PROTECTED] wrote: What do you think of this project www.openpbx.org ? Something like ser and openser ! Interesting. In their meeting minutes (http://wiki.openpbx.org/tiki-index.php?page=Meeting+Minutes+10-5-2005) I see that a BKW was elected to the board. Is this Brian West? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: www.openpbx.org
Further info. The domain is registered to Marc Olivier Chouinard. He has posted in the dev list. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: www.openpbx.org
gincantalupo [EMAIL PROTECTED] wrote: why a fork??? I don't know any of the people involved, or what their motivation might be, but I will make a guess: Digium's model tends to stifle innovation. Look at eclipse.org for a much better model. Eclipse is truly open source. IBM's commercial products are built on top of Eclipse. No parallel licensing scheme. No restrictions on what can go into the project as a result of trying to maintain the dual licensing. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Fax Problems
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] wrote: Wonder if anyone would (or has) write a fax gateway app that would read it off the PRI or whatever then store and forward. Would enable 'internet faxing' without the requirement for T.38 or similar. There is already a spec for this. T.37. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Any work around for ISPs that block port 5060 and 69
Asterisk guy [EMAIL PROTECTED] wrote: it seems bindport setting doesnt work. any idear on how to change the sip listening port ? port= Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Loop Detection
Daniel Corbe [EMAIL PROTECTED] wrote: Is there any way to turn Loop Detection off or tune the params a bit? I am having an issue with Call Forwarding on my SIP Proxy Server which is causing me great pains. All I can do is sympathize. The same problem occurs when a call comes in through Asterisk, gets sent to SER, then comes back to Asterisk 20 seconds later for voicemail. I have contemplated just commenting out the check in chan_sip.c, but I haven't tried this. Not sure if this might cause other problems. Asterisk has many SIP deficiencies. Asterisk has been built as a monolithic PBX, and it seems to do okay using SIP phones as channels. If you want Asterisk to simply act as a SIP UA, you are going to run into a whole slew of problems. I'm not holding my breath waiting for this to change. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Beeps during Sip to Sip phone calls
Eric Wieling [EMAIL PROTECTED] wrote: Daryll Strauss wrote: Yep, I've seen it and from reading http://www.voxilla.com it's a pretty common problem. If you turn on debugging what you'll see is that the Sipura has mistakenly detected a DTMF code in the audio stream and is relaying it by repeating the signal (very loudly I might add) So this appears to be a bug in the most current firmware. I've reported it to Sipura including the debug output. Maybe more people should do the same. You'd think that switching to RFC2833 DTMF would fix that. That is actually the problem. It thinks it hears DTMF so it sends an out-of band signal. The other end receives this and produces the audible tone. Switching to in-band fixes it. Well, works around it. :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura 841 issues
Master Abi [EMAIL PROTECTED] wrote: Not having a backlit display is bad design. Actually it is a feature issue, not a design issue. :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: diffrent area codes for diffrent phones in dialplan
Jer [EMAIL PROTECTED] wrote: I have 3 sets of SIP phones all in diff area codes that need to access the PSTN I need to it so that a 7 digit number is converted to a 10 digit with the correct ara code eg a call coming from sip-phone1 needs aera code AAA and a call coming fom sip-phone2 needs BBB how can this be setup in the dialplan is there someway to set a var on a per sip group basis? I thought of the accountcode...since i will not be using it for CDR How about a different initial context for each area code? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: voicepulse silence during conversations
Sean Kennedy [EMAIL PROTECTED] wrote: Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. I have noticed this too, especially when speaking to someone who is using a cell phone. I assume that VP is using silence suppression. -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream Message button
Peter Bowyer [EMAIL PROTECTED] wrote: Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog. Is this a beta version of the firmware? The main download page only has 1.0.5.16. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: iax2-jitter-trunking?
Mark Eissler [EMAIL PROTECTED] wrote: AFAIK, trunk=yes is not a global option. You set it within a context. Also, using the jitter buffer with trunk=yes is not recommended since its broken right now. The jitter buffer itself is broken, or only in combination with trunking? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Pro biz Asterisk
Frank Kostin [EMAIL PROTECTED] wrote: What about modifing or adding some extra code sources, ... etc? The GPL requires that if you make changes, you make the changed source available to anyone to whom you distribute the changed application. You needn't make the source publicly available, but you can't restrict those to whom you provide it from doing so. If your changes are used internally, you haven't distributed the changed application, and there is no issue of having to make the source available to anyone. If you are providing a service using the changed code, you still haven't distributed the changed app. (everybody duck now) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: different IAX ports for different contexts
dean collins [EMAIL PROTECTED] wrote: My question is this, can you have different ports for different contexts within IAX? If I understand correctly, you want to use different *local* ports for different contexts. I don't think you can do this. You could run more than one copy of Asterisk on the same machine, and configure each one to use a different port. -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BRI in the US?
Michael Graves [EMAIL PROTECTED] wrote: I see rates for BRIs in the state tariffs here in Texas. When I speak with SBC they are willing to sell them, but say that they are usually installed for pure data applications where DSL in not available. The rates seem comparable to POTS if you consider calling features extra on POTS lines. Here in New Brunswick, Canada, BRI is available to us. It costs about 50% more/line than POTS. Caller-ID, and other calling features are not included in this price for either BRI or POTS. As you can guess, it was difficult to find someone able to provide this information. -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Mediatrix 1204 (4x FXO)
Christian Hecimovic [EMAIL PROTECTED] wrote: Also, we couldn't get ringback to work right, despite a correct indications.conf file - if someone calls in and dialed an extension, then the ringing they hear is very choppy and messed up. We had this problem. Solved it by disabling silence suppression on the Mediatrix. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: res_motv: Request for Comment
Mark Spencer [EMAIL PROTECTED] wrote: I've been considering the nature of Asterisk, its security, the bug tracker, and more... And i've come up with an interesting idea: A message of the version. The idea is that Asterisk has a compile time [...] a) The idea itself -- is it a good one or is it stupid? It could be a useful feature *if* done right. Some other people have already made some good comments on this. I think it seems like somewhat of a waste of time with the number of truly useful things that could be done to Asterisk instead. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Disambiguating incoming IAXTel calls
Brian Cuthie [EMAIL PROTECTED] wrote: I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. When I researched this issue a few months ago, I found a post from Mark Spencer saying that the dialed number was intentionally not sent to discourage people from setting up multiple IAXTel numbers for the same server. In a mailing-list discussion a month or so ago, somebody said that he was getting this information in the calls, but had to specifically request a configuration change by Digium. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Spring VON Wrap Up
Scott Laird [EMAIL PROTECTED] wrote: On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? It is only SIP that would be on TCP. RTP (media stream) would still be UDP. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream and codec G.711
Mireia Munoz de jesus [EMAIL PROTECTED] wrote: My gateway accepts G.711, but not my Grandstream 100 series SIP phone Mine does. It is termed PCMU and PCMA in the Grandstream setup. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: can't logon to voice mail - bad password
Paul Mahler [EMAIL PROTECTED] wrote: I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 = 3213,Bill Smith Did you solve this yet? Maybe you have a non-ASCII character in the file. Try deleting the line and retyping it. Or cut and paste a working entry and modify it. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AGI crashes asterisk
Vikram Rangnekar [EMAIL PROTECTED] wrote: I configured agi-test.agi on extension 111 when i dial into asterisk extension 111 using a IAX softphone and hangup while the AGI is playing asterisk crashes. Does anyone have any idea why this happens. I encountered this same problem using Red Hat 9. The problem was resolved by setting an environment variable ASSUME_KERNEL or something like that. The machine has been re-tasked, so I can't check the variable name, but you should be able to find it in the archives. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100P Echo was: USB Headsets (Plantronics DSP-400)
Jason A. Pattie [EMAIL PROTECTED] wrote: Is there any possibility to remove the turnaround leg or whatever its called at the X100P? I'm just thinking of a scenario where none of the outgoing signal is ever introduced to the incoming circuit. That way, the echo problem simply disappears. Or is this not possible? What if the X100P were modified to do this in hardware? Go ahead and let it balance the circuit if that is necessary, but then not introduce any of the outgoing signal onto the incoming circuit. I believe that some high-end hardware does this. The ADIT 600 claims to. I wish that the X100P did, as we have three that we are unable to use due to echo. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sip proxy
Thomas B. Clark [EMAIL PROTECTED] wrote: I have tried working around by setting up my own DNS to be authoritative for provider.com, and providing a SRV record with a proxy in it, but Asterisk is ignoring it. There is a sip.conf option to tell Asterisk to do SRV lookups. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: USB Headsets (Plantronics DSP-400)
Ed Rubright [EMAIL PROTECTED] wrote: I'm thinking about getting the Plantronics DSP-400 headset for use with Xlite softphone. I currently have a analog headset that does NOT have a DSP on board, which gives me mediocre call quality and echo when talking to the PSTN thru my X100P card. I have zero echo when talking thru my X100P on my cordless phones attached to the Digium TDM400P. Before I got spend the money I was wondering if others using USB headsets with a DSP and getting good results? My thought was thought by using a headset with a DSP on board the echo would go away? Probably not. The echo is occurring on the PSTN. You can't hear it through the TDM400P because the latency is so low that you don't notice it. When you move to VoIP, the extra latency lets you notice the echo. You need to eliminate this echo at your connection to the PSTN. That means trying to cancel it using the Zap options for this. We didn't have much luck doing this for our X100Ps. It appears that the echo is actually being caused on our local loop. Probably an impedance mismatch between the X100P and the line. This occurs on three different CO lines with three different X100Ps, at two different sites. We have ordered a Mediatrix SIP/PSTN gateway. Hopefully this will solve the problem. We will probably stick the X100Ps in a closet somewhere to keep dust off the shelf. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Message waiting indicators
Oliver Wilcock [EMAIL PROTECTED] wrote: A post in 2002 refered to Mike Sandman as a source for inexpensive (cheap) message waiting indicators. I called Mike but he doesn't know what Asterisk is (!) and wants to know what type of phone system I have or what protocol it uses so that he can send me a compatible indicator. I tried these acronyms on him: ADSI, MDMF, SDMF but he doesn't recognize them. What can I tell him so that I can order the right part? Or which popular switch is Asterisk compatible with? I guess it depends on how you want to hook this to Asterisk. If though a TDM card, just tell him it is POTS. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAXTel multiple registers?
John Fraizer [EMAIL PROTECTED] wrote: And what I told you works just fine. I'm taking 20+ DIDs from a single IAX provider with no problems what-so-ever. I'll be happy to consult for you at my normal hourly rate if you still can't figure it out. That is different. Your provider sends the called number identification. IAXTEL doesn't. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI and a SIP ATA
We are interested in deploying some ADSI phones, but we are currently using Sipura SPA-2000s. Everything I read on ADSI, says that it is just audio, and you don't need a special channel bank or anything like that. But what about through the SPA? I notice that zapata.conf appears to have an option to turn on ADSI support. This makes me think that it won't work with any other channel. I searched the mailing list, but the only thing I could find was from Oct. 2002, at there was no definitive answer. Anybody know if it will work? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: No Ringback Tone when Dialing (outside caller to internal extension from auto-attendant)
Paul Vermette [EMAIL PROTECTED] wrote: Well, I think I discovered even further why there is no ringback tone available. The following message, is displayed on the console in asterisk. ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Looking more into it, I found that it was related to loading tones for a particular zone. The message is printed out when I run the Ringing Application or when I attempt to put the 'r' option on the Dial application in my extensions.conf. I don't know if this helps, and you may already know this, but tones for regions are defined in indications.conf. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Anybody know about the Sayson 480i VoIP Screen Phone?
Steven Sokol [EMAIL PROTECTED] wrote: In looking for a screen phone to use with Asterisk I came across Sayson's new Aastra 480i SIP/MGCP/H323 screen phone. It looks just like the ADSI phone but offers dual 10/100 Ethernet jacks and apparently some kind of programmability much like the ADSI phone has, but using XML. Before I go pestering the nice people at Sayson, I thought I would see if anybody here knew anything about it - including details on the licensing/locking scheme. Also, any guess as to the price would be cool. I'm guessing it will be more than the equivalent ADSI. I have been watching this phone for two or three months now. I don't think it is actually available yet. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New to T-1/Channel Bank hardware -- help?
Rob Fugina [EMAIL PROTECTED] wrote: I'm considering a small office setup with at least 12 extensions. Seems (as has been stated in previous threads) that for the FXS ports, a T100P and a channel bank could be the most cost-effective way to do this. I'm not so sure about that. 6 Sipura SPA-2000s will cost you about US $540. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Garbled Faxes
Jim Sneeringer [EMAIL PROTECTED] wrote: My incoming and outgoing faxes are garbled and sometimes get disconnected. I remember reading somewhere that I should use u-law for faxes, but I don't know how to do that. The fax is connected to a Digium FXS card and the calls come in or go out on a Digium FXO card. Can anyone tell me how to fix this? If the FXS and FXO cards are in the same box, then I don't think there is any issue with codec selection. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re-Invites and Studder.
Billy Huddleston [EMAIL PROTECTED] wrote: I've been using Asterisk with a Cisco GW and ATA's.. I have it setup with re-invites. When a call is first answered, the 1st second or so of the conversation is stuttered, garbled, whatever you want to call it.. I believe this is due to Asterisk shifting the media stream directly to the Gateway or ATA. Is their a way to eliminate this stutter without disabling re-invites? This is very discontenting to our customers and employees... Why don't you turn off re-invites as a test and see if the problem goes away. Then you will know for sure if this is causing the problem. Or to put it another way, the first step in solving a problem is identifying the problem. :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: incoming call to internal user
David J Carter [EMAIL PROTECTED] wrote: Matteo, try: - [incoming] include = default ;default location for internal phones exten = s,1,Answer exten = s,2,Wait 10 exten = s,3,Dial(SIP/100) exten = s,4,Hangup I don't think that will work. There is no mention in the documentation that Wait will accept DTMF. You want to set a timeout and then on timeout (t extension, I think) dial the fall-back number. Prior to the timeout, the user will have a chance to enter an extension. See the manual for more details. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk-grandstream call
Bill Michaelson [EMAIL PROTECTED] wrote: I am trying to muddle my way tthrough getting something - actually anything to work - with Asterisk. I've acquired a Grandstream phone and I've got * on a Red Hat 9 box. I've gotten to a point where I can see (via ethereal) that the phone REGISTER's successfully with asterisk, and then I try to dial into voicemail. This is what I observe in the packet trace... GS: INVITE - * *: Status 100 (Trying) - GS *: Status 200 (OK with session description) - GS Does the GS then send an ACK? It should. If it doesn't then this probably means that it hasn't received the 200 response. (firewall issue?) If it is sending the ACK, then it is probably a codec issue, as has been already mentioned. GS doesn't always seem to do very well in codec selection. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP error
Kannaiyan Natesan [EMAIL PROTECTED] wrote: Means RFC3389 support is incomplete. Neither Mark or developers @ digium will not accept it when it gets completed by anyone. I'm not sure what you are saying here. Do you mean that if someone completes support for this, that Digium will not accept it for inclusion in Asterisk? If this is what you mean, why won't they accept it? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Junk calls from FWD numbers
Chris Albertson [EMAIL PROTECTED] wrote: Now, I'm noticing that I have both Sue Albertson and Chris Albertson listed and it is Sue who gets 3x as many of these calls. And the callers are all men, I suppose? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream 100 sidetone
dkwok [EMAIL PROTECTED] wrote: For people who are using GS 101, what do you think the sidetone generated by the phone. Seems fine on the two we have. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. Interesting, we don't experience this. Are you sure this is side tone and not a far-end echo? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inter-Fone (was Mediatrix 1204 sip experience?)
Jess Magnaye [EMAIL PROTECTED] wrote: Go for inter-fone products. it can both support sip and h323. I took a look at their site, and some of the products look quite interesting. The prices don't seem too bad either, if I am interpreting them right. It is hard to tell if the prices include the FXS/FXO modules or not. Interesting (at least to me) observation number 1: they have a US address, but their copy appears to have been written by a person with a first language other than English. Observation number 2: I did a quick web search and couldn't find much of anything about them. No re-sellers even. Perhaps they only sell direct? Has anybody actually used any of these boxes? What did you think? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call
Chris Wilson [EMAIL PROTECTED] wrote: Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Just guessing here, but it sounds like Asterisk sent a request, didn't get a reply, sent again, didn't get a reply, and so on until it hit an internal limit. If my guess is correct, I suppose there could be many causes, including: * Target host down * No path to the target * Firewall blocking traffic * Target host not running SIP, at least on the targeted port. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SayDigits
Chris Wilson [EMAIL PROTECTED] wrote:
Has anyone had this problem:
(When calling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446 ast_openstream: File digits/ does
not exist in any format
Jan 24 10:50:27 WARNING[-1252262992]: file.c:734 ast_streamfile: Unable to open
digits/ (format ULAW): No such file or directory
in Extensions.conf
exten = 1010,1,SayDigits(${CALLERID})
/var/lib/asterisk/sounds/digits exists, and there are many files in there. Any idea's?
File permissions?
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
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[Asterisk-Users] What technology could my phone company be using?
I live in New Brunswick Canada. The phone company is Aliant. When you set up business service here, you can go with either analog or digital lines. This isn't a T1 or ISDN. They are talking individual lines direct to handsets that they provide. They offer the digital option with even very small ( 2 - 4) number of lines. What technology could this be? Is there any way to connect such a line to Asterisk? PBX vendors that I have talked to in the past say that we can bring the Aliant digital lines straight into their PBX. We would like to do this with Asterisk, but can't figure out what the technology is. (ever try to get technical info from the phone company?) I have searched the web and the archives, but can't figure out what the technology could be. It can't be T1, because these lines come direct to handsets. I don't believe it can be BRI because each phone has only a single line, and the price seems to low. If anyone can shed any light on this, it would be very much appreciated. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP and AGI crash...
I experienced exactly the same behavior as described in the original message. When running an AGI, Asterisk would crash if the caller hung up, but not if the script completed and the dial-plan did the hang up. This only occurred with Asterisk running in daemon mode. Nicolas Gudino [EMAIL PROTECTED] wrote: Are you running RedHat 9? If you do, try with this line before launching asterisk (with stock redhat 9 kernels): export LD_ASSUME_KERNEL=2.4.1 I am running RH 9 and this seems to have solved the problem for me. Thanks. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Power Over Ethernet for *any* ethernet switch (or hub); product idea
Stephen R. Besch [EMAIL PROTECTED] wrote: As far as I know, it doesn't. The POE source somehow monitors the line (using impedance, etc) to determine if there is anything connected to the pairs used to supply power. Since netcards and net connected equipment are not supposed to use the power pairs, this should work in most cases. If the termination just leaves them unconnected the impedance is infinity and power will not be applied. Conversely, if the termination are all grounded, impedance is (near) 0 and power will not be applied. For some range of impedances in between, the far end is assumed to require power and power will be applied. So a reasonable (cheap!) alternative would be to only plug devices that need PoE into it, and have it always provide power. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What technology could my phone company be using?
Jon Pounder [EMAIL PROTECTED] wrote: I live in New Brunswick Canada. The phone company is Aliant. When you set up business service here, you can go with either analog or digital lines. This isn't a T1 or ISDN. They are talking individual lines direct to handsets that they provide. They offer the digital option with even very small ( 2 - 4) number of lines. I am guessing DSL to a hybrid box like one of the ones from ZHONE that split off several pots lines or isdn lines, and a high speed serial line (normally configured as frame relay for whatever bandwidth is left over) This box could be at your demarc much like you would have your T1 modem (T1 is actually delivered over HDSL on a single pair, and then the modem box splits it out onto the the familiar tx and rx pairs) In this case the handsets could be either analog or isdn depending what flavour box they provide. They could also be just giving you an adsl circuit, and a dumb modem/hub, and providing unique ips to IP phones that you happen to plug in, but there would not be any network connection to the internet, just to their voip headend at the CO. I think this technology pre-dates DSL and maybe even ISDN. It is also deployed in large organizations with hundreds of phones. I think they use this for Centrex. It seems to use RJ-11 connectors and flat wires, if this means anything to anyone. I believe the phones are Nortel. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What technology could my phone company be using?
Mark Hazlewood [EMAIL PROTECTED] wrote: Sounds like Centrex services, we had it from Telus in Alberta a few years ago. I believe this is used for Centrex. I thought Centrex was basically a CO-hosted PBX. Is it also a local-loop technology? Are there PCI cards or SIP gateway boxes available? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Digium X100P for $43
SamW [EMAIL PROTECTED] wrote: Digium X100P / new cards are is available on ebay for $43. Actually they seem to be made by Digit Networks. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How to diagnose pops and clicks?
Peter Rukavina [EMAIL PROTECTED] wrote: My setup is as follows: Handset - Sipura SPA 2000 - Asterisk - VoicePulse and Handset - Sipura SPA 2000 - Asterisk - Digium X100P - POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant popping and clicking on the line. This isn't enough to interfere seriously with the call, and the voice quality is otherwise telephone quality. People I'm calling to don't report hearing the pops and clicks on their end. I'm looking for advice as to how to best diagnose this problem. Interesting. Last night I placed a call: Budetone 102 - Asterisk - VoicePulse I periodically noticed pop/click followed by a brief silence. The person I was speaking to couldn't hear this. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
