Re: [asterisk-users] zaptel telephone cards and asterisk in another pc
All of those lines going to desktops must be fed from someplace, like a wiring closet maybe? Why not put your FXO gateway there instead of at the desktop where your users will break them? On 2/20/09, Paul Hales pdha...@optusnet.com.au wrote: Why do you need so many Asterisk installs? With the ability of Asterisk to handle hundreds of lines/phones/etc, the need for several Asterisk server is generally for very specific situations. PaulH Ignacio wrote: Jeff I will take a more depth look at those linksys devices this weekend but I think they could be very interesting. Tzafrir, what I like to avoid is installing an asterisk server in every user computer. I think that is useless I want only one server to mantain. On Fri, Feb 20, 2009 at 7:55 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Fri, Feb 20, 2009 at 07:11:04PM +0100, Ignacio wrote: Thank you very much for your fast answer Eric. I was trying to avoid to have to install as many asterisk as pcs I have. But I think there is no way to do it. I only have seen something like network block device, but not sure if it is going to work and quite difficult to configure properly. Anyway I think the fast and easier way will be installing and asterisk in every client. I guess you can use TDMoE. But I'm not really sure it will give you a lower overhead. Specifically, why is it that you want to avoid installing Asterisk there? The requirements of an Asterisk system for a few analog channels and a few uncompressed SIP/IAX channels are rather minimal. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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 to Avaya
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN signalling all sorted one way, and can pass
calls from the real world (ie. the telco and asterisk) TO the avaya
box, and it accepts that and sets up the call perfectly.
The problem is that the Avaya box is signalling outbound calls using
an odd method, which smacks of an analogue system with ISDN30 bolted
on for a bit of a laugh.
They send a q931 SETUP message. This contains the correct callerID,
but only the first 1 to 4 of the dialled number's digits - The
remainder of the number is I believe passed through using DTMF!!! From
the look of it they intentionally do not send an IE 161 Sending
Complete with the SETUP, so that the far end continues to listen for
these DTMF tones, until it resolves to a legal number.
My questions for some ISDN expert out there...
Part 1) I need to receive the number in the SETUP, which I guess will
be in ${EXTEN}, then I assume I can use Read() to collect DTMF digits,
and check the dialplan to see if it is a locally terminated number.
Once I am 100% sure it is not local, I can then dial the collected
number through the Telco ISDN channel. Make sense? I think I can
probably handle that. The problem is that I do not know whether I have
received all digits from the Avaya at that point, which leads to...
Part 2) Can I dial through Zaptel (via a Sangoma card if that makes a
difference) without sending the IE 161 call complete? I thought that
Dial(Zap/G1||D(${INITIAL}))
might send the initial digits using DTMF, and then leave the channel
open so that more DTMF could follow over the now bridged channel. In
fact I get an immediate failure as if the far end thinks I have
finished dialling. Can I assume that libpri does not currently support
this method of dialling? If not, how might it be added? I can hack the
code, I just need suggestions of where to look and how it might sanely
be added :)
Part 3) It is possible that the Avaya is not using DTMF at-all, and
that it will send more bits of the called-party number using the
D-Channel as you would expect, but Asterisk does not seem to be
waiting for them. Can this be changed in Zaptel/Asterisk. Does anyone
know the Avaya systems well enough to suggest how it might be working?
Many many thanks for any feedback.
Regards,
Steve
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Re: [asterisk-users] SellVOIP
Well, my $21 is still there and all of my calls are being declined. Over a year ago, I requested a refund and regardless of all promises that I would receive one, Jed never followed through. I'd use up the credit if the calls would only complete. On Sat, Apr 5, 2008 at 1:03 AM, Ira [EMAIL PROTECTED] wrote: At 09:42 PM 4/4/2008, you wrote: Common practice is to check every bill. Withing the last month, I have found two several hundred dollar mistakes on Credit Card and Checking account. I am nos sure if companies are charging extra to make up for the economy slow down or they are genuine mistakes, but I have never had these issues in the past besides a mistake here and there over the course of a year.. Good advice, but that wasn't what my message was about. They're a VOIP provider that I thought went out of business months ago or maybe a year ago with $13 of credit on my account. Today they re-appeared and my $13 is still there. Likely that's true for others too. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ata device but for a soundcard
This may be a good place to start looking: http://www.atlassound.com/index.cfm On 2/20/08, Jerry Geis [EMAIL PROTECTED] wrote: I am looking for an ATA like device but instead of VOIP to analog phone I want VOIP to low level audio out. Something that looks like a sound card output. I know I can use cheap PC's but that then you have HD's to setup etc... HD failures etc... Anyone know of something like that? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya 9620 phone using Firmware 2.0.1.34 has working MWI lamp
I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk 1.2.22), using Avaya SIP Firmware version 2.0.1.34. Set [EMAIL PROTECTED] in the sip.conf Found MWI worked immediately. Turned off as expected. Have Fun! Tom ___ -- 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] Definity G3R and MWI
You're looking for Leave Word Calling activation and deactivation. On 12/28/07, Doug Lytle [EMAIL PROTECTED] wrote: Henk Dick wrote: Doug, Have you checked the feature access code that is defined in the definity. That is the code that needs to be dialed. I always checked the codes from a definity phone to make sure that I was using the right I have not been able to find any references to the feature codes available for the Definity G3R. The Definity manager wasn't able to locate any documentation either. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
I suspect if you remove the callerid entry from this device's sip.confdefinition things will work better. On 9/9/07, Apa Minerala [EMAIL PROTECTED] wrote: I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call 778f89593967725f0abe40eb1752504c no reply to our critical packet. What is the critical packet that is not being responded to? Please help. -- Pinpoint customers http://us.rd.yahoo.com/evt=48250/*http://searchmarketing.yahoo.com/arp/sponsoredsearch_v9.php?o=US2226cmp=Yahooctv=AprNIs=Ys2=EMb=50who are looking for what you sell. ___ 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 ___ 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] Avaya SIP phones (4610SW) and MWI
I'd love to hear about this as well. On 7/27/07, Derek Fedel [EMAIL PROTECTED] wrote: Hi all, I'm new to the list, so I apologize in advance if I'm beating a dead horse by asking this, but I read somewhere that asterisk 1.4 has MWI working for Avaya and their rather troublesome SIP firmware. Can anyone verify this before I go changing phone systems around? Thanks Derek ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.)
On the other hand, the guy could just be using his work e-mail for personal interests. On 7/7/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: On Sat, 2007-07-07 at 08:39 -0500, [EMAIL PROTECTED] wrote: Date: Fri, 06 Jul 2007 12:02:53 -0600 From: Stephen Bosch [EMAIL PROTECTED] Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Wayne wrote: I was wondering where 3Com were getting all the new ideas from for their phone system ;-p Cats out of the bag now I guess :) The price of open source is that the commercial outfits are free to rip off ideas without paying for them. But hey -- competition is good, right? Competition is good, one benefit of OSS pressure on commercial/proprietary competitors to improve their products which lead investment. Cooperation is also good. Public knowledge that corporations are in the community helps us know where to look for GPL software they secretly use, or just how they get some valuable ideas from which they profit (profit from us, usually). So it's easier to convince them to explicitly feed back into the OSS. Either just user feedback, or actual investment in testing, further development, or even GPL'ing their own proprietary tech into the community. So now it's time that 3Com hears from us, and we hear back, that we're all coopeting together. If they don't explicitly contribute soon, that bad community attitude will be a clue for some examination of their products for included GPL code and GPL violations, or just some bad press for being merely takers with their $billion budgets. -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 ___ --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] really strange behavior
exten = _X.,1, On 6/3/07, BSumrall [EMAIL PROTECTED] wrote: Understood, it is not the catch all but, what if I am designing a server that needs to accept calls from 15 or more 1800 numbers? How would you now channel it to a catch all? Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Sunday, June 03, 2007 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] really strange behavior In short, the 's' extension is not a catch-all. The use of 's' can be confusing. The best example I have of the use of 's' is when a ZAP call comes in on an analog line. IIRC, the book says something to the effect that 's' is for when, upon arrival in a context, the call has no other place to go. Works for me :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: RE: Digital Phones
Bilal, I don't think anyone is telling you that digital phones don't need cards. I do think they are telling you that NOBODY makes a card that drives digital phones for use with Asterisk. Your initial assumption that Digital phones work with Asterisk is untrue. They can be made (forced?) to work using a CITEL gateway, which communicates with Asterisk via SIP and ethernet, but only for phones from a handful of manufacturers. Previous posts cast doubt on the results you can expect. I don't know how to say this any more clearly. I too am Avaya certified and I have no illusions about using Avaya or Nortel digital phones with Asterisk. On 5/10/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; As I know from AVAYA (I am AVAYA certified) that digital phones are connected to digital cards and it does not go through ethernet switches at all, digital phones should be independent on the ethernet network, so if the network down, these phones will start working, it will be totally isolated from the data traffic. So, how that come the digital phones does not need a card for it? Also, how it will use ethernet switches! It does not work with IP Packets. Any one can advise. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?
At the very least, he's abusing his customers. Substances? I hadn't thought of that. On 4/30/07, Salvatore Giudice [EMAIL PROTECTED] wrote: I suspect that Jed has a substance abuse problem and that he may be in rehab. I don't know for sure of course. This kind of silence is indicative of people being hauled back to rehab. Anyway, maybe he just makes a habit of running off with people's money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Servers
You could also look at Oreka at sourceforge. On 4/13/07, Matthew J. Roth [EMAIL PROTECTED] wrote: Savoy, Kevin - Williston, ND wrote: We are looking at using Asterisk as a call recording server for an Avaya VoIP S8700 system in a multi-site VoIP Call Center. All calls will be coming in to one location and sent out via VoIP to other call centers. What kind of specs should we be looking at purchasing for our Asterisk server to be record up 200-300 calls simultaneously? I can tell you from experience that disk I/O will be your bottleneck. In our testing, call quality began to seriously deteriorate at around 60 simultaneous calls. Our solution was to record to a RAM disk and move the leg files over NFS to a dedicated server for mixing, indexing, and retrieval. It has been a while since the tests, but as far as I know app_monitor based call recording still has a disk write in the code path that bridges two channels. Unless another kind reader of this list can provide updated information or a better call recording method, I'd assume this is still the case. For our inbound call center operations, we regularly record roughly 200-300 simultaneous calls on a single server. The agents and queues also reside on that server, but we have taken great care to offload as many processes as possible. We aren't 100% stable, but I believe most of our downtime can be attributed to app_queue. If no new information surfaces, I'd be happy to talk with you about dimensioning and our overall architecture. Keep in mind that if its an option, breaking this task down so that it can be spread across multiple machines will minimize the number of headaches you're going to have. Linux runs in 64 bit architecture, but does Asterisk actually take advantage of the 64 bit? That really depends on how you define take advantage. We are running on a 64-bit architecture, but I have no idea if we're any better off because of it. From what I've read, you typically have to benchmark processes to see if they are faster on a 64-bit OS. One definite advantage of 32-bit is most code is more heavily tested for it. Some assumptions that programmers might make, such as casting a pointer to an int, do not port very well to 64-bit. However, if you end up going down the RAM disk path, you might want to research whether or not a 64-bit OS would be necessary to provide enough memory. Has anyone tried doing this already? What would be the best way to get the calls from the Avaya PBX over to the Asterisk recording server? Any thoughts? I haven't implemented this particular case, but off the top of my head I would say that you could register the Avaya PBX as a SIP user agent. Then you could direct all of the calls to the Asterisk server which would utilize dialplan logic to record them and bridge them to their desired endpoints. Once again, I'm relying on the other readers of this list to point out any naivety on my part. My best thinking doesn't usually occur at 8:30 on Friday evening. I hope this is helpful, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - IP Network Call Recording
Check out Oreka at sourceforge, too.(aka OrkAudio) On 2/15/07, Kristian Kielhofner [EMAIL PROTECTED] wrote: On 2/15/07, Cory Andrews [EMAIL PROTECTED] wrote: Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and allow an administrator to monitor or recording inbound/outbound calls. If anyone is aware of a mechanism or solution that would provide this capability, please shoot me an email. Thanks Cory Andrews Cory, From their website it appears they are using SIP. With any luck it will be SIP + ulaw (without re-invites). If so, do this: 1) Get a decent managed switch that can setup monitor ports. Configure one port to monitor the port connected to the Talkswitch. 2) Get a decent dual-homed machine. 3) Connect one interface of the dual-homed machine to the monitor port. Running Linux, do an ifconfig up [interface name] (no IP address). Configure the other interface to connect to a network for management, copying files, etc. 4) Start up tcpdump on the interface, writing to a file. 5) Use something like Cain + Abel to read the RTP and dump the audio to a file. 6) Convert files to desired format using sox. The only step I left out was Profit!. Seriously though, this depends on a few key assumptions about the Talkswitch: 1) That it is standard SIP. 2) It uses ulaw. 3) It doesn't do re-invites. Not any one of these is a show stopper for this type of sollution, but any one of them (or all of them) could make life a bit harder for you... Good luck! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy and MOH
Lacy, it appeared to me that he was calling himself an idiot. Thanks for some of the background on the issue, though. On 3/27/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: WOW that fixed it! What an Idiot. I was going somewhere with that, but never mind. Good luck. Maybe the idiot is the guy who posted no additional details of his configuration, in particular, whether the CLI was showing music on hold starting, and then stopping, or whether the music on hold process was continuing but no sound. If it was a timing issue, by rubbing your hand across the mouthpiece, I would guess it is generating interupts for the timer to work and music on hold works, until you stop rubbing it and it fades it out. Hitting or tapping the mouthpiece produces the same outcome. Or, it that doesn't produce anything, it could be a permissions problem. It could be something not configured correctly in the config file. It could be that you are using mp3s instead of native format, as Andrew had asked about. But, since I'm an idiot, what do I know? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?
Balu, I suspect the author was expressing sarcasm. On 3/26/07, Balu Raman [EMAIL PROTECTED] wrote: Can you tell me, why sellvoip rocks ? I am looking to sign up with someone. Thanks, balu raman On 3/25/07, Stephen Bosch [EMAIL PROTECTED] wrote: Salvatore Giudice wrote: Nothing has changed in my Asterisk configuration and now outbound US is getting nothing, but 403's. Anyone else having the same problem? Inbound calls to my DID's are working fine. Clearly, sellvoip rocks! -stephen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?
I'm not surprised. On 3/25/07, Stephen Bosch [EMAIL PROTECTED] wrote: Salvatore Giudice wrote: Nothing has changed in my Asterisk configuration and now outbound US is getting nothing, but 403's. Anyone else having the same problem? Inbound calls to my DID's are working fine. Clearly, sellvoip rocks! -stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Refund from SellVoip?
I, on the other hand, have been disappointed repeatedly by their failures to route international calls. I've received e-mails from them promising a refund. I expect them to keep their word. On 3/24/07, Martin Joseph [EMAIL PROTECTED] wrote: On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said: Now I know where they've been spending my remaining balance... I still use Sellvoip as my primary terminator, and have found the call quality to be superior to any other ITSP from my location (Seattle). I agree completely that there is no support from this company, which is a major issue if you are trying to support other customers. Still, I remain a happy customer of sellvoip, with Teliax and Nufone configured as backups... I wouldn't expect a refund for cancellation of prepaid phone usage, does the original agreement you have with then suggest that they owe you a refund? Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refund from SellVoip?
Now I know where they've been spending my remaining balance... On 3/21/07, Ira [EMAIL PROTECTED] wrote: At 09:08 AM 3/21/2007, you wrote: Does anybody know Jed Stafford? As far as I can tell this ended up being a one-man or two-man operation. It's just sad. I got a marketing email from them last week telling me about all their cool new features. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refund from SellVoip?
This is probably why they don't use PayPal anymore. Now, there is no resolution process that I can pursue, other than complaining to the Gov't., which I have. On 3/20/07, Vicky [EMAIL PROTECTED] wrote: I got money back around 6 months ago . It was a via paypal claim and hey didn't reply till paypal's deadline so i got $30 back . On 17/03/07, Ira [EMAIL PROTECTED] wrote: At 02:32 PM 3/16/2007, you wrote: You were able to cancel service with Sellvoip? That's impressive, that Actually, it's Voxee I tried to cancel and failed. I still use SellVOIP and it mostly works but support is a problem. I'm starting to use using Telasip more though as they work and have a POP only 19ms from here, a big advantage. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Refund from SellVoip?
Has anyone been successful in getting a refund from SellVoip when you've cancelled service? Tom Lynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refund from SellVoip?
At this point, I'm simply contacting the State of Washington Attorney General's office. They're ignoring my e-mails and I'm done monkeying around. On 3/16/07, Ira [EMAIL PROTECTED] wrote: At 11:32 AM 3/16/2007, you wrote: Has anyone been successful in getting a refund from SellVoip when you've cancelled service? No, I'm just using the credit up slowly whenever their network works. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7912
Do they appear to have failed as a result of Daylight Savings time? On 3/14/07, Matt Putnam [EMAIL PROTECTED] wrote: I didnt have them on tftp files they were all manualy configured. They are not trying to request anything they have the tftp server address but are not requesting any files. It should start up and look for a vlan but its not even doing that it does nothing when i plug it in just a blank screen and the red and green leds on the hold and menu buttons are lit. On 3/14/07, Hermann Wecke [EMAIL PROTECTED] wrote: Matt Putnam wrote: anything useful any sugestions? Are they requesting anything via TFTP? Do you have the full tftp files ready? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anybody having problems using sellvoip?
International calls (Germany) haven't completed since around 3/1. Domestic works. Is it just me? I'm getting 503 responses. Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: AsterikNow vs Trixbox
The question sounds like troll bait to me. On 2/11/07, Edward Halman [EMAIL PROTECTED] wrote: In the beginning... I tried both. I found Trixbox to be a very effective out-of-the-box solution for those venturing into the world of Asterisk IP PBX without wanting to learn about dial plans. I was able to install it and configure it with my VoIP provider and SIP phones in about 2 hours. FreePBX is a great web interface for the novice. If it weren't for my company's more advanced needs (like AGI), I'd probably never have bothered to go further. I also tried AsteriskNow, out of curiousity mostly. I was unable to configure it with its web interface to work with my VoIP provider. I found the documentation to be lacking as well. I have (thankfully) grown out of the need for such front ends and have (thanks in no small part to O'Reilly and this list) learned how to develop my own dialplans and write my own configuration files to get rid of all the unnecessary stuff that comes with Trixbox. I have version 1.2 running on FC5 and couldn't be happier. So if you don't want to learn Asterisk, I believe Trixbox is the way to go. But ultimately, learning to configure Asterisk on your own is well worth the trouble. Edward Halman (718) 705-7451 [EMAIL PROTECTED] -- Date: Sun, 11 Feb 2007 11:21:15 -0800 From: [EMAIL PROTECTED] Subject: [asterisk-users] AsterikNow vs Trixbox To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Comments? People's opinions -- Thanks http://www.sqlhacks.com The SQL knowledge base ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] Happy 2007!!!
Dovid, you're killing me. This after asking if we can't all just be nice to each other. On 1/1/07, Dovid B [EMAIL PROTECTED] wrote: Adam and bill are both wrong. The world revolves around me. Geeez cant we cut the crap (i.e. Happy new year is followed by a response that hey it isnt the new year here yet) If you need the attention find a place where there is a live TV feed (report) and say I am a tool, I need attention Geez.. (As a disclamer don't do it. I just hope you get my point) - Original Message - *From:* Bill Hackensack [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Monday, January 01, 2007 6:08 PM *Subject:* Re: RE : [asterisk-users] Happy 2007!!! On 12/31/06, Adam Jacob Muller [EMAIL PROTECTED] wrote: It's still 2006 here -Adam Well, Adam, I guess it is all about you. What does the rest of the world look like as it revolves around you? -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya to Asterisk via H323
Andrew, in your experience, what has changed from version to version. I work daily with Avaya gear and do regression testing on new releases before they hit the public. In my experience, I can't recall anything changing with h.323 trunking other than the maximum number of trunk members managed by a signalling group, and that was a while ago. On 12/29/06, Andrew Latham [EMAIL PROTECTED] wrote: Mark I would start with setting up two asterisk boxes and configure an H.323 link between them, then as you have it working as you like bring the Avaya into the fold. that way you know that 50% of your settings are done (bound interfaces, settings and the like). From what I remeber Avaya may change setups from version to version. I am looking forward to tackling this on a 1000+ multi site in the neer futrure, what fun Andrew On 12/29/06, Mark Greene [EMAIL PROTECTED] wrote: I am tasked with linking an Avaya Definity switch to an asterisk box using it's IP card that handles H.323. All my googles turn up a lot of results but nothing recent. I am able to find instructions but they are dated from 2005, and often fail halfway through. What is the best way to achieve what I want, which is two way calling between the Avaya switch and Asterisk server using h.323, and where do I need to look for setting it up on centOS 4.4? Thanks in advance, - Mark ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Happy 2007!!!
Sounds like an EBay ad... On 12/30/06, Josué Conti [EMAIL PROTECTED] wrote: Always... Desire that in the New Year that if you really initiate... It hears the words that always it desired to hear. It pronounces the phrases that one day it desired to repeat. It feels the emotion that always waited to feel. It walks for the tracks that one day it desired to follow. It divides the affection with who always desired to distribute. It hugs all the friends whom always it desired to congregate, and alive the life that always dreamed to exist... Happy 2007 Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
I agree, he sent me one off list, too - making all kinds of allegations of my sexual preferences. I sent him a link to AA, DrPhil, National Institute of Mental Health and suggested he get some help. On 12/28/06, Alex Robar [EMAIL PROTECTED] wrote: Of course everyone is allowed to use VoIP... Asterisk is open! I think Dovid's point was more that this guy's website says he buys and sells precious metals and other random items, his postings on this list indicate that he installs PBXes and resells VoIP services, and then his private e-mails say that he's a PI. The PI thing sounds just like him trying to get those who attacked him to back off. Alex On 12/28/06, Kevin Walsh [EMAIL PROTECTED] wrote: Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/ [EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Here's what he sent me after I told him to shut the up. I kind of wonder if he's just trying to generate traffic at certain sites and it's going to generate ad revenue for him in some lame scheme. Oh well: So you are one of the scammers you are dog shit Good bye you are now blocked like Steve is You are a want to be some one like Bent! Are you in bed with him? Most be I guess you two are good butt buddys :-D :-P Get a life asshole and stop trying to become a geek Your site is slow and looks link shit my dog could do better and he can't type. Get it on a real hosting and get it off your cable/DSL Internet connection You don't have the brain power. 1st graders have more than you do. If you don't want the links to scams then you can't handle the truth Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email http://www.bochterservices.com/?t=Email On 12/28/06, Alex Robar [EMAIL PROTECTED] wrote: As if we needed more proof that Bochter was a screw-ball... He's now accused me of being the owner of TRXTel. Not that we needed proof he wasn't actually a PI, but in case anyone had any doubts, read the thread. Alex -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Dec 28, 2006 7:41 PM Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: Alex Robar [EMAIL PROTECTED] There are small minded then there is you Bent Fuck you Your spoof email address is blocked Get a life and stop your scams by hiding.. use a real email address... You are a waste of my time GOOD BYE :-P Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Alex Robar wrote: If you actually wanted to give the information to people, you would have just posted it instead of ranting like a lunatic. Your real problem is that you need attention. Stop being a diva and deal with stuff like this on your own. The bottom line is that if you actually had a case, you would have just proceeded with it and dealt with this privately like any normal, decent person would have done. My gut tells me you have jack shit in terms of evidence, and you were just fired as a customer by Brent for pulling shit like this... Something I would certainly agree with him on if that's what he did. I'll bet this never moves forward and we'll never hear anything about any action you've taken. In fact, I'll bet we'll see the inverse - That TRXTel has sued you for libel for attempting to defame them in public. And FYI, I actually did answer your question, you just didn't read my response... Something quite common in your responses, it seem. Alex On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex But if you READ the posts. I replied to all OFF THE LIST So that is YOUR POINT... They posted my replys That were off the list to the list I blocked the other two jackasses on the server to stop there pointless messages. They can't send any messages to any users at any domains on my servers. The same as we are talking OFF THE LIST // The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. What you need to do is check into the PERSON (*Thats one owner*) that is around 28 years I have a list of 32 others that were scammed by bent Ask me for the links on textel no one as asked for the links.. The point is I am not going to waste any more of my time on the ones like you that don't what the information on the truth. *By the way you never answered my question Do you want to be scammed and lose your money???* New question?? What is unlimited use So your replys are pointless Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Alex Robar wrote: The POINT that you keep whining and complaining about so much, is that you're trying to bully and scare people into ceasing their posts that reflect negatively on you. The original points of your post are not what anyone is focusing on anymore - YOU moved the points away from that by insulting people. Everyone else who is off the point is simply responding to you. The issue here is not that anyone LIKES to be scammed... But that you've insulted valuable, respected members of the Asterisk community simply because of a bad experience you had. To post a complaint is one thing, to rip into someone the way you did is quite another. The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. Alex Robar On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex This is off the list. The point is that I don't like scammers. The ones that tried to attacked are some of the scammers that I am dealing with. Do you like to get scammed out of your money? And what is the
[asterisk-users] Toll-Free number in India
Can anybody point me to a vendor that can provide a toll free number that can be used in India to reach the united states? Verizon Business is telling me they can't get one. Thanks - Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
And it seems likely to me that you'll be sued for libel. On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED][EMAIL PROTECTED]wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
I second that. I'm quite happy with the IPKall.com did number I use today. Only once in the last year was it unavailable when I needed it. So, not bulletproof, but good enough for me to use all day when I work at home. On 12/21/06, www.IPKall.com [EMAIL PROTECTED] wrote: One way audio is almost always caused by firewalls / NAT translation. Since there is neither on IPKall, my guess would be to look at the other end. With 20k + users, most have succeeded in correcting this problem via their hardware / software. I encourage you to look at the user forum for some suggestions. IPKall IPKall Forum http://voxilla.com/PNphpBB2-viewforum-f-38.html -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Al Bochter *Sent:* Wednesday, December 20, 2006 4:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Need quality toll free 800 number over IAX? I have used www.ipkall.com I have had one way audio for two weeks now with no reply from CS. So I will back you up on this I guess http://www.kall8.com/ would be the same I think they are one in the same. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Kevin Walsh wrote: www.IPKall.com [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Anyone except NuFone. Their customer service is non-existant - you have to email every day for a couple of months before you'll be privileged enough to get a one-line response to a service outage issue. If you dare to point out that the response didn't address the issue then you'll unleash the combined wrath of both of the brain cells in residence at NuFone's support department. Not immediately, of course - you'll have to wait another couple of months for a reply. If you give up on them and decide to go elsewhere, they will pocket any outstanding funds you have pre-paid into your account. Existing NuFone customers are advised to not pre-pay too much to these yokels, and to jump ship as soon as possible. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5.8gig phone MWI
You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. On 12/8/06, Doug Crompton [EMAIL PROTECTED] wrote: Thanks, but unfortunately that is an expensive 2 line phone compared to others in their line that have a base and two or three remotes for the same price. Seems a lot to pay for a MWI. I wonder if anyone has had experience with panasonic wireless 5.8gig and MWI?? They advertise compatibility on some models but I also saw a review comment that it did not work. Doug On Fri, 8 Dec 2006, Steve Prior wrote: Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on a Home rotuer
It may not be what you're thinking, but I use Astlinux on an older PIII. With a couple of options it has become my home router and works very well. On 12/7/06, Dovid B [EMAIL PROTECTED] wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk connection to a PBX
How many channels do you require? I'd favor T1 for a few reasons. Higher port density means fewer cards per system, which will mean fewer interrupts. T1s won't require you to tune analog levels. Echo probability will be lower. On 11/29/06, asterisk-robert [EMAIL PROTECTED] wrote: We are thinking of setting up an Asterisk system to route calls between 2 of our factories. Our idea is to connect an Asterisk box to each PBX and then use SIP(or IAX) to truck between the 2 systems on our internal network. I would be interested in any ideas regarding the connection points: 1. Is using Asterisk a good solution? 2. Is using a T-1 card the best way to connect the PBX and Asterisk? 3. If analog is used for the connection is it better for Asterisk to use FXO or FXS cards? Any ideas are appreciated. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Port 5060
Can you tunnel through a VPN connection? On 11/28/06, Patrick [EMAIL PROTECTED] wrote: On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote: We have many clients who live in third world countries where the ISPs purposely block traffic on port 5060. I know we could always change the listening port in our Asterisk box. However, doing so will affect all our other users who use port 5060 with no problems. Is there any other solution? I guess I could always run a second instance of Asterisk listening on another port, but is that the cleanest and most scalable solution? Have you tried redirecting the other port with iptables to port 5060 on the Asterisk box? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best text to speech program
Cepstral sounds good and it's cheap. However, it still sounds like a synthesized voice. On 11/28/06, Hall, Eric M. [EMAIL PROTECTED] wrote: I'm looking to set up asterisk to call customer 3 days before the app and remind them we will be out to see them. I'm looking for any ideas on good ways to do this. Also I think it would be best to do some type of text to speech however I do not like the sound of the free one . Any ideas? Thanks!!! Eric Hall ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Survey: In what ways do you use Asterisk at your house?
Earle, I'm running Astlinux on a PIII 550 with 384 megs of ram. Booting from a Compact Flash card. Non-Volatile storage on a USB Keydisk. I have three SIP DID numbers in three different area codes here in Western Washington via IPKall. I use a local SIP termination provider and also retain my Qwest POTS line with callerid, which connects via an X100P clone board. All in all, I've got less than $50 sunk into the system. Past that, I added an SPA3K, which is an utter disappointment, regardless of what old Ward Mundy has to say about them. I'm got various services running. Weather reports via an cepstral speech synthesis (runs off-server in a wmware instance). I use it to block unwanted callers from reaching my home based on values stored in the ASTDB and simple dialplan logic. Speed Dials, wakeup calls, music on hold customized to the caller based on Caller ID, Conference bridge (remember, I've got three DIDs via my broadband), and I'm also working on automating retrieval of e-mail and conversion to speech so that my in-laws, who don't have a computer, can hear their e-mails as soon as they arrive, via the telephone, subject to common sense time of day rules. I have DISA service setup for the rare instance that I get an urge to call overseas from my cell phone. And when they build FAX capability into AstLinux, I'll use that, too. On 11/23/06, Neil Cherry [EMAIL PROTECTED] wrote: Earle Clubb wrote: - What service provider/technology do you use for origination/termination? - What hardware/software do you use and how does it all tie together? - What tasks do you use * to accomplish? - Any other pertinent info. Until last summer I had Asterisk doing the normal call handling my home. You know selecting which line to call out on via an SPA-3000 and SPA-3102. We do have trouble with the SPA's as the echo can be quite bad or the volume is quite low (take your pick). I'm also routing various calls to various vm-boxes and sending selected callers to the SIT. I also had an extension that interfaced to Mr. House home automation software. I could control and monitor a few things in my home. This system is no longer working due to a drive crash and the lack of backup for parts of this setup. I'm hoping to get the time towards the end of the year to put it back together. I may try to integrate the voice recognition (Sphinx) into the setup also. This was running on a 1GHz/512M/300G vanilla x86 clone. I had printer services, DNS, DHCP, file sharing, home automation, Asterisk and a few other things running. It's also my development system. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Rewriting caller ID from database?
Vincent, I do something similar to what you're doing. However, I use the CID number as the astdb family, allowing me to assign multiple attributes as the keys. It requires some maintenance, so I also wrote a php script for the management. You can find it here: http://voip-info.org/wiki/view/Web+based+Asterisk+Database+maintenance Family: 206-456-7890 Privilige: 1 Family: 206-456-7890 Music:jazz Family: 206-456-7890 Recording: No By inverting the relationship, I found it easier to focus on the source of the call and the treatments I want to apply. I can also wipe out entries by family name and remove all attributes in one operation using database deltree. On 11/25/06, Time Bandit [EMAIL PROTECTED] wrote: I use some custom scripts to do database lookups and rewrite CallerID information. Everything works fine with regard to the CID name, however my Cisco 7960 and Linksys SPA-942 phones do not display the calling number. Instead, they display the called number. This makes the phone's call return feature not work. The calling number and name are both properly displayed on all of the softphone clients that I've tried. Here's the format I'm using to set the CallerID. SET CALLERID JONES DARYL A6508701826 If you're using Asterisk 1.2, see this page : http://www.voip-info.org/wiki/view/Setting+Callerid hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto recording calls?
quicktime player does it without adding any codecs.
On 11/24/06, Tim Panton [EMAIL PROTECTED] wrote:
On 21 Nov 2006, at 14:34, Jay Moore wrote:
Tim Panton wrote:
On 20 Nov 2006, at 21:46, Jay Moore wrote:
Doug wrote:
Hmmm. I think this may work for WinAmp and
incidently for Windows Media Player:
http://www.mlkj.net/gsm/
No luck with WMP. Anyone else have any suggestions on
playing .gsm files in Windows Media Player?
Jay, would you be interested in a java applet that played gsm files ?
I think I have the bones of one kicking around that I could dust off
and polish up.
This only really works if you are providing your customers access
to the gsm files
via http and can easily wrap a page around them...
Well, ideally I'd like for my customers to be able to download the
file and play it on their computer, but a Java applet that plays
them on our website would be a cool idea, too. So, yeah, I'd be
interested.
Here's my bare bones implementation GSM player for voicemail etc
http://www.westhawk.co.uk/software/playGSM/PlayGSM.html
As the web page says, you can use it in 2 ways:
1) as an applet - arrange for the web app (php?) to set the 'url' param
and it will play download and play the selected file
2) or if you download the jar file (http://www.westhawk.co.uk/
software/playGSM/PlayGSM.jar)
you can also run it as an application
java -jar playGSM.jar {url of gsm}
It is GPL - so enjoy and fix bugs
Tim.
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Re: [asterisk-users] Re: Re: Rewriting caller ID from database?
I like a challenge. I'll let you know if I come up with anything. On 11/26/06, Vincent Delporte [EMAIL PROTECTED] wrote: At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote: By inverting the relationship, I found it easier to focus on the source of the call and the treatments I want to apply. I can also wipe out entries by family name and remove all attributes in one operation using database deltree. Interesting. I'll give it a shot tomorrow at the office. Anyhow, at this point, I could successfully import all the name + number records, and must find solutions for the following problems: - web interface to add/modify/remove records - find out if LookupCID is able to match prefixes with a record (some of customers have DID, so I'd like to just use 123-45?? to match those incoming calls to Such and such customer instead of adding individual records 123-4501, 123-4502, etc.) Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Questions . . .
Jason,If you must stick with analog phones, you can find higher density channel banks that will host 8, 16 or up to 24 ports each. They communicate back to your asterisk server via your LAN. Or, as has been stated, you can purchase IP phones that also communicate back to your asterisk server via your LAN. This will leave you dealing only with the FXO ports. If you look at Sangoma gear, you can probably achieve what you're looking for and only occupy 1 PCI slot (even though the card needs the space of two PCI boards). On 11/13/06, Sharon Lim [EMAIL PROTECTED] wrote: Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php . Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck On 11/14/06, Jason Flatt [EMAIL PROTECTED] wrote: Hello all.My company currently has an older Executone PBX system that we are outgrowing.Rather than wait until the last minute to make a hasty decision, I thought itwould be a good idea to do some research and compare options first.My expertise is in computers and networking, and telephony systems are mostlyforeign to me.What we currently have are 5 incoming POTS lines and 25 stations and arewanting to add 1 or 2 more stations.I think we might have added at least one more incoming line, except that the phones we have only support 5 lines(so I'm told).Our PBX system has room for 5 more stations, then it's timeto buy a new one.I'm assuming I need to add some hardware in order to make Asterisk work with our existing setup, but I'm not entirely sure what.Based on the readingI've done so far and my limited understanding, if we wanted to use it inplace of our existing PBX system, I would need to get an analog interface card (several, actually), like Digium's TDM400P, like so:2 - Wildcard TDM04B cards for FXO and7 - Wildcard TDM40B cards for FXS-or-1 - Wildcard TDM04B card for FXO and1 - Wildcard TDM22B card for FXO FXS and 7 - Wildcard TDM40B cards for FXSI might as well use the top configuration for future expansion.If I am correct, that is 9 PCI cards in a PC.I don't know of any motherboardthat supports that many cards, so either I'm wrong, or I'll need different cards, or I'll need to utilize 2 or more PCs in conjunction with each other.I haven't yet found any mention on the last two options, so I'm assuming I'mwrong and I need a little enlightenment.Thank you for any information that will help me better understand this. --Jason FlattFather of Six:http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis, 9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005)Linux User: http://www.sourcemage.org/Drupal Fanatic: http://drupal.org/___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Definity Asterisk CallerID Issue
Steve poses some good questions. In addition, I'd wonder how your trunk group in the definity is configured? Are you sending calling party name and number? If so, is your DS1 card set for protocol A,B,C, or D? For both calling name and number, I believe you need B, but I don't have my docs here. If all that works out, then I'd move onto your ISDN public numbering table. You should administer this table to define the format of your calling party number.On 11/5/06, cp [EMAIL PROTECTED] wrote: I am hoping someone could shed some light and point me in the right direction? I'm attempting to get callerid work between an Avaya Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the Definity side I've searched endlessly and came with an example which we modeled as close as we can, but still no luck. While doing PRI intense debug span 1 in I see a couple interesting messages but have yet to come up with meaningful knowledge about them. I've tried decoded the setup message but don't know what I'm really looking at. It appears in the decode that the calling party number or name are not being sent but as I mentioned I don't know what I am really looking at. I wondering if these error messages have any thing to do with Asterisk not knowing what to do with what the Definity is sending? Feel free to contact me offlist. Any assistance is greatly appreciated. -CP !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Progress Description: Calling equipment is non-ISDN TON: International Number xxx*CLI [ 02 01 d4 d2 08 02 0a e6 05 04 03 90 90 a2 18 03 a1 83 8b 1e 02 81 83 70 05 91 34 33 38 39 96 08 04 d0 35 30 80 ] Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 2790/0xAE6) (Originator) Message type: SETUP (5) [04 03 90 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a1 83 8b] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 11 ] [1e 02 81 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [70 05 91 34 33 38 39] Called Number (len= 7) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '4389' ] [96] Locking Shift (len=01): Requested codeset 6 [08 04 d0 35 30 80] !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Sending Receiver Ready (107) [ 02 01 01 d6 ] ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxee lag problems ?
Regardless, they're still perpetually lagged. I'm suspicious as to why paypal is conducting a review. For now, considering the poor performance, I stand by my decision to shop the market. On 11/11/06, Vicky [EMAIL PROTECTED] wrote: I doubt how many days more voxee will survive . Its been a month nw and tehir support doesnt want to repair this . ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxee lag problems ?
Then I guess I'd better hurry up and use my remaining 49 cents worth of credit!!On 11/11/06, Vicky [EMAIL PROTECTED] wrote:I doubt how many days more voxee will survive . Its been a month nw and tehir support doesnt want to repair this . ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundfiles adding during phone calls
Ron,The guy is trying to help you. Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording. The best part of it is that you can hang up and the recording will continue to play to the caller. When it expires, so does the callOn 11/11/06, Ronald Wiplinger [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: http://www.voip-info.org/wiki-Asterisk+config+features.conf... and where exactly did you see this feature byeRonald Wiplinger On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I want to add some sound filed on demand during a phone call only possible on some extension numbers. I get many phone calls from local companies, but don't understand Chinese! I would like to record the call, but also ask the caller some questions, which should be added into the call with some keys on the phone, ... e.g.*66554 should add into the call: How are you? or What is your phone number? But I do have another application for that too. I get many fake phone calls, where Chinese people tell you that your phone bill is not paid, your court fee is not paid, and ask the caller to go to the ATM machine and key in a series of key strokes, most likely it will clear out your account. For such fake callers I would like to add a terrible noise to the call and make scare them as much as possible. Such fake calls I get now for each of my phone lines at least 10 each!!! Either the caller-id is not set, is 0 or is a tollfree number. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0647-0, 2006/11/09 Tested on: 2006/11/11 �U�� 11:07:21 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com --Ronald Wiplinger(CEO of ELMIT)http://www.elmit.comhttp://voip.elmit.com http://e-paper.elmit.comTel. (M) +886.939.775.516(O) +886.2.2835.7765 (ENUM) or FWD 511208- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention!Our system is protected with a spam prevention program.If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please.After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxee lag problems ?
Add me to the list. Not only lagged, but also failures to register. AND, apparantly Paypal won't automatically authorize payments to them anymore. I'm not recharging my account anymore. On 11/10/06, Tim Panton [EMAIL PROTECTED] wrote: On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote: Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two...Now there is a definitive case of a 'lagged' communication channel! :-)Tim Pantonwww.mexuar.netwww.westhawk.co.uk/___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on GUI
Without providing a link to the list, or citing your front-runners, you can't really expect people to reply, can you?On 10/27/06, Frédéric Blaise [EMAIL PROTECTED] wrote:Hello allI would like to know your opinions on free GUI used to manage Asterisk. Which is better?My setup is quite small, about 15-20 phones. I've seen the liste onvoip-info.Thanks all.fred___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Embedded platforms
astlinuxOn 10/25/06, Prasad Kandikonda [EMAIL PROTECTED] wrote: We are looking at porting asterisk onto a embedded platform based on IXP or ARM. I would like to know the experiences of anybody who has already ported to these platforms. I am also particularly interested in issues related to performance and scaling on these platforms. Also, is anybody aware of any embedded asterisk products. I know recently Digium announced a platform based on Blackfin.Thanks, Prasad Kandikonda. Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
Are your sip phones capable of auto-answer?I can imagine you can terminate the incoming call into a meet-me conference (no pass code) and then trigger a script that creates a call file for each of the other participating phones. The auto-answer part seems like the sticky part. On 10/15/06, Marc Heckmann [EMAIL PROTECTED] wrote: On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote: The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally.In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital NortelMeridian phone. The one phone has to be analogue because it interfaceswith a radio broadcast phone patch.-m Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for a Voicemail Lamp device
I'm looking for an external device that can flash when there is new voicemail in a mailbox. I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system. Problem is, the Uniden system has it's own answering machine, which I don't want to use. But the message lamps are driven solely by the internal answering machine function. Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime.Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
My uniden phone is the TRU8885-3HS.On 10/14/06, Bob Chiodini [EMAIL PROTECTED] wrote: Tom,The uniden TRU446 and the CLX465 both are supposed to detect stutterdial tone (SDT) from the phone company and light the MWI.When used with asterisk the SPA3000 can generate SDT.I'm not sure it can do soon its own.I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCUhttp://www0.epinions.com/content_70491344516Hope that helps, a little. Bob...Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox.I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system.Problem is, the Uniden system has it's own answering machine, which I don't want to use.But the message lamps are driven solely by the internal answering machine function.Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
I can get stutter dialtone using my spa3000, but the uniden doesn't respond to it by lighting the lamp. All it sees is an incoming call from the spa.It looks to me that I'll either need an external MWI device or I'm going to have to replace the Uniden phones. On 10/14/06, Tom Lynn [EMAIL PROTECTED] wrote: My uniden phone is the TRU8885-3HS.On 10/14/06, Bob Chiodini [EMAIL PROTECTED] wrote: Tom,The uniden TRU446 and the CLX465 both are supposed to detect stutterdial tone (SDT) from the phone company and light the MWI.When used with asterisk the SPA3000 can generate SDT.I'm not sure it can do soon its own.I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCUhttp://www0.epinions.com/content_70491344516 Hope that helps, a little. Bob...Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox.I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system.Problem is, the Uniden system has it's own answering machine, which I don't want to use.But the message lamps are driven solely by the internal answering machine function.Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How do you like TrixBox?
FWIW, I too started with AAH, but got really upset when tempted with an upgrade and learning the path was a total re-install. I hear things have gotten better since.In response, I went completely minimalist and turned to AstLinux. My primary reason was my only hardware resource was a PC without a hard drive. It cost me 1/2 as much to buy an IDE to Compact Flash adaptor and a 512MB CF card than a new hard drive. With my dusty old driveless PC now converted to a brand new, yet still dusty * system, I turned to the O'Reilly book, freely downloadable from the net. Working through the book, I learned everything I needed to configure a beginning system. I now run my home on AstLinux, and regardless of how much I hear about TrixBox being upgradeable, I still see it's a love/hate relationship every time a new release is put forth. I also read a lot about audio quality and AstLinux gets great marks in this respect. I'd recommend it to anybody new to * because it's minimalist, embedded systems approach teaches you so much about the tradeoffs you have to make to maintain a stable system that pleases it's users. TomOn 10/14/06, Michael Collins [EMAIL PROTECTED] wrote: I first learned asterisk via [EMAIL PROTECTED] Then I went to straight asterisk.This seems to be a theme.Getting your feet wet with [EMAIL PROTECTED]/Trixbox is nota bad way to go, especially if you want to get a functioning system up and running quickly.After tinkering with Trixbox then go back and do aplain Asterisk install.You will learn a lot, both about Asterisk andTrixbox.I've modified the Trixbox install scripts a bit to tailor them to my needs and ended up with the best of both worlds: a Trixboxinstallation that is more than plain vanilla but less than the somewhatcluttered Trixbox stock install.-MC___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipSupply? [Semi-Urgent] (Big Apology)
I've never ordered from Voipsupply, but did forward two questions to their President. The first one was trivial and off-topic, yet answered very quickly. The second was product related and answered equally quickly and knowledgeably. I'd definitely consider purchasing from them, based on my experience.On 10/13/06, Shaw Terwilliger [EMAIL PROTECTED] wrote:Shaw Terwilliger wrote: If you search the archives from a few months ago you'll find a few unhappy voipsupply customers (including me).They never shipped what I ordered, didn't respond to any e-mail or calls.The president saw the list traffic and sent me a long apology (stating his commitment to service) and offered to send me an extra component that I had cancelled the order for--free of charge--as a show of good will. It's been two or three months since that promise, and I never received the part.He hasn't responded to my follow-up did you really mean it? e-mail either.I must offer a HUGE apology to VoipSupply in regards to my first reply. VoipLink.com, *NOT* VoipSupply, was the company I had problems with (as described in my first message).Except for sending me some spam after Iordered from them, I have had no problems with VoipSupply.I confusedthe vendors as I wrote my reply, since I have ordered from both of them. Sorry for the confusion, and best wishes to the VoipSupply team.--Shaw Terwilliger [EMAIL PROTECTED]SourceGear LLC___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya 8300 - Asterisk integration using H.323
I'm told the 4.0 release of communications manager will support up to 100 SIP phones without the need for an extra server. Are you trying to connect stations or trunks?On 10/13/06, Andrey Kovalenko [EMAIL PROTECTED] wrote: Hi everyone,I was wondering if anyone on this group has successfully integrated Avaya 8300 or 8700 and Asterisk using H.323 trunk and would be willing to share configurations and/or comment on the voice quality achieved. Currently we have Avaya 8300 integrated with Asterisk over a Q.SIG trunk, but we need to put Asterisk in a different geographical location from the PBX andneed to explore other options.Thanks. Andrey___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3102
Dave, Are you in the US? On 10/12/06, Dave Cotton [EMAIL PROTECTED] wrote: On Thu, 2006-10-12 at 21:30 +0200, Csibra Gergo wrote: Thursday, October 12, 2006, 6:58:57 PM, Tim wrote: I've read alot of comments on the SPA-3000, many if not all saying they had echo issues, but I've not seen anyone comment on the SPA-3102. Does anyone have any comments or issues with these? Well, I have had echo issues. Then I find out the echo cancellation on PSTN line is switched off by default. I switched on, and no echo any more :)I have had echo with the SPA3000 but I switched to Global impedance on the FXO and since then clear as a bell.--Dave Cotton [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] e911
I'm keeping my Qwest line for this purpose.On 9/23/06, Christopher Corn [EMAIL PROTECTED] wrote: Im using voipestreet and voxee for my SIP termination. neither of them, offer any kind of e911 service. as i search the web i see different companies that offer this e911 service to voip suppliers. I want to choose the right one, seeing how in an emergency, it can be very crucial. any suggestions? thanks. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Operator Console(s)/Shared Call Appearances
That's simply the remaining rationalization that is left in the absence of the bridged line appearances. On 7/25/06, Matthew Warren [EMAIL PROTECTED] wrote: Subject: [asterisk-users] Re: Operator Console(s)/Shared Call Appearances Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk and I'm trying to determine the best way to allow our receptionist to answer certain executives telephone lines. It seems there are probably two routes, but I'm not sure of the limitations of each. You could make both the executive and the receptionist phones ring,perhaps with a very low ring tone for the executives. Then thereceptionist will take the call whenever possible. If the call needs to go through to the executive, the receptionist can do a direct calljust by pressing a button, and a different (perhaps louder) ring tonecan play.This is what call park and call pick-up groups are for.Exspecially if you use an operators panel.Matt___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
perhaps not what you're looking for, but reading thru your config, it looks like you've mis-spelled 'echo cancel' as 'echo cancle'On 7/23/06, Frank Darner [EMAIL PROTECTED] wrote: What is the output from 'cat /proc/zaptel/*' After delete of all Asterisk files and complete new install I got something: # modprobe zaptel modprobe wcfxo linux:/proc/zaptel # cat /proc/zaptel/* Span 1: WCFXO/0 Generic Clone Board 1 RED1 WCFXO/0/0 but # ztcfg - is still no channels. any ideas?OK, I have now an output.For some reason ztcfg was not looking in /etc/asterisk for the zaptel.conf.After using the option -c /etc/asterisk/zaptel.conf everythink was fine. One last thing:Playback sounds now like MickeyMouse, much to slowalso Calls SIP to SIP are not any more possibleany ideas?___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with sip debug?
Rich,I had the same problem and the solution was to take out a 'malformed' callerid value from my sip.conf entry.TomOn 7/20/06, Rich Adamson [EMAIL PROTECTED] wrote:Tried the syslog debug, but it reports the exact same thing as the sip debug shown below. It includes the INVITE, 100 TRYING, AND 486 BUSYHERE. There are no hints as to why the Busy Here message is returned.I was kind of guessing that something in the sip header was not as expected for the device, but I don't see anything that seems to beinappropriate in the sip debug.Thoughts?Shanon Swafford wrote: I always like to activate the syslog and debug on my SPA's.Sometimes this will tell you what they are doing. Shanon -Original Message- Need a little help trying to understand what's happening here. spa941 - Asterisk-A - iax2 - Asterisk-B - spa942 When the spa941 (x3000) calls spa942 (x1004), the spa942 returns a busy here sip message. The spa942 is not busy and does not have DND or any other option set to cause a busy-here message. Asterisk-B is v1.2.10 updated to current svn. (Seeing the exact same issue with an spa3k.) A sip debug from Asterisk-B shows the following three packets: localhost*CLI sip debug peer 1004 SIP Debugging Enabled for IP: 160.80.40.201:5060 == x1004-- Registered IAX2 to '151.213.193.101', who sees us as 153.222.216.140:1963 with no messages waiting-- Accepting UNAUTHENTICATED call from 151.213.193.101: requested format = gsm, requested prefs = (g726|gsm|ilbc), actual format = g726, host prefs = (g726|gsm|ilbc), priority = mine-- Executing Dial(IAX2/to-npi-3, SIP/1004|15|r) in new stack We're at 160.80.40.4 port 13382 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 160.80.40.201:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe;rport From: NPI-Rich sip:[EMAIL PROTECTED];tag=as0e37bb22 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 19 Jul 2006 22:27:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 261 v=0 o=root 18182 18182 IN IP4 160.80.40.4 s=session c=IN IP4 160.80.40.4 t=0 0 m=audio 13382 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - - Called 1004 localhost*CLI -- SIP read from 160.80.40.201:5060: SIP/2.0 100 Trying To: sip:[EMAIL PROTECTED]:5060 From: NPI-Rich sip:[EMAIL PROTECTED];tag=as0e37bb22 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe Server: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (8 headers 0 lines)--- localhost*CLI -- SIP read from 160.80.40.201:5060: SIP/2.0 486 Busy Here To: sip:[EMAIL PROTECTED]:5060;tag=e434eff616a11501i0 From: NPI-Rich sip:[EMAIL PROTECTED];tag=as0e37bb22 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe Server: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 486 Busy Here back from 160.80.40.201 Transmitting (no NAT) to 160.80.40.201:5060: ACK sip:[EMAIL PROTECTED] :5060 SIP/2.0 Via: SIP/2.0/UDP 160.80.40.4:5060;branch=z9hG4bK544dbabe;rport From: NPI-Rich sip:[EMAIL PROTECTED] ;tag=as0e37bb22 To: sip:[EMAIL PROTECTED]:5060;tag=e434eff616a11501i0 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 - SIP/1004-081e9c08 is busy == Everyone is busy/congested at this time (1:1/0/0)-- Executing VoiceMail(IAX2/to-npi-3, 1004|ug(6)) in new stack-- Playing 'vm-theperson' (language 'en') Destroying call '[EMAIL PROTECTED]'-- Playing 'digits/1' (language 'en')-- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en')== Spawn extension (from-sip, 1004, 2) exited non-zero on 'IAX2/to-npi-3'-- Executing Hangup(IAX2/to-npi-3, ) in new stack == Spawn extension (from-sip, h, 1) exited non-zero on 'IAX2/to-npi-3'-- Hungup 'IAX2/to-npi-3' In addition, if I access the spa942 via a web browser, all lines/extns are idle. Does not seem to be any reason for the 'busy here' message that I can see.Placing a call to another spa942 on the same Asterisk-B and on the same wire works fine.Yesterday the first spa942 was working fine as well. Can anyone see anything strange in the sip debug that would cause this? R.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --
[asterisk-users] 489 Bad Event
I'm trying to debug why the message waiting lamp on my phone won't light. I suspect it doesn't adhere to standards.I used tcpdump to capture it's bootup sequence. From the dump, I can see the phone trying to subscribe to my asterisk server ( 1.2.7), to which it receives an initial 401 unauthorized response. On the second notify request, it receives a 489 bad event response.Below is the second notify and the 489 bad event response. I'm hoping that someone who is more skilled than I can see what is wrong with the notify request. TIATomNo. Time Source Destination Protocol Info 148 52.196856 192.168.1.50 192.168.1.1 SIP Request: SUBSCRIBE sip:[EMAIL PROTECTED]Frame 148 (713 bytes on wire, 713 bytes captured) Arrival Time: Jul 14, 2006 18:30:45.344749000 Time delta from previous packet: 0.046162000 seconds Time since reference or first frame: 52.196856000 seconds Frame Number: 148 Packet Length: 713 bytes Capture Length: 713 bytes Protocols in frame: eth:ip:udp:sipEthernet II, Src: 192.168.1.50 (00:04:0d:50:22:8e), Dst: 192.168.1.1 (00:b0:d0:5e:1e:a6) Destination: 192.168.1.1 (00:b0:d0:5e:1e:a6) Source: 192.168.1.50 (00:04:0d:50:22:8e) Type: IP (0x0800)Internet Protocol, Src: 192.168.1.50 (192.168.1.50 ), Dst: 192.168.1.1 (192.168.1.1) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 00.. = Differentiated Services Codepoint: Default (0x00) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 699 Identification: 0x0041 (65) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0xf46d [correct] Good: True Bad : False Source: 192.168.1.50 (192.168.1.50) Destination: 192.168.1.1 (192.168.1.1)User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 679 Checksum: 0xede0 [correct]Session Initiation Protocol Request-Line: SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Method: SUBSCRIBE Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bKecd8ca820 Max-Forwards: 70 Content-Length: 0 To: 203 sip:[EMAIL PROTECTED] SIP Display info: 203 SIP to address: sip:[EMAIL PROTECTED] From: 203 sip:[EMAIL PROTECTED] ;tag=7463f27760ed517 SIP Display info: 203 SIP from address: sip:[EMAIL PROTECTED] SIP tag: 7463f27760ed517 Call-ID: [EMAIL PROTECTED] CSeq: 1903799861 SUBSCRIBE Route: sip:192.168.1.1 ;lr Supported: timer Expires: 86400 Event: message-summary Contact: 203 sip:[EMAIL PROTECTED]:5060 Contact Binding: 203 sip:[EMAIL PROTECTED]:5060 URI: 203 sip:[EMAIL PROTECTED]:5060 SIP Display info: 203 SIP contact address: sip:[EMAIL PROTECTED]:5060 Supported: replaces Authorization:Digest response=8533130fceedb3d6c8f14ea63c79d278,username=203,realm=asterisk,nonce=66fd9409,uri= sip:[EMAIL PROTECTED] User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 00 b0 d0 5e 1e a6 00 04 0d 50 22 8e 08 00 45 00 ...^.P...E. 0010 02 bb 00 41 00 00 40 11 f4 6d c0 a8 01 32 c0 a8 [EMAIL PROTECTED]0020 01 01 13 c4 13 c4 02 a7 ed e0 53 55 42 53 43 52 ..SUBSCR0030 49 42 45 20 73 69 70 3a 32 30 33 40 74 6f 6d 6c IBE sip:[EMAIL PROTECTED] 0040 79 6e 6e 2e 63 6f 6d 20 53 49 50 2f 32 2e 30 0d ynn.com SIP/2.0.0050 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 .Via: SIP/2.0/UD0060 50 20 31 39 32 2e 31 36 38 2e 31 2e 35 30 3a 35 P 192.168.1.50:50070 30 36 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 060;branch=z9hG40080 62 4b 65 63 64 38 63 61 38 32 30 0d 0a 4d 61 78 bKecd8ca820..Max0090 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 43 -Forwards: 70..C 00a0 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 68 3a 20 30 ontent-Length: 000b0 0d 0a 54 6f 3a 20 32 30 33 20 3c 73 69 70 3a 32 ..To: 203 sip:200c0 30 33 40 74 6f 6d 6c 79 6e 6e 2e 63 6f 6d 3e 0d [EMAIL PROTECTED].00d0 0a 46 72 6f 6d 3a 20 32 30 33 20 3c 73 69 70 3a .From: 203 sip:00e0 32 30 33 40 74 6f 6d 6c 79 6e 6e 2e 63 6f 6d 3e [EMAIL PROTECTED] 00f0 3b 74 61 67 3d 37 34 36 33 66 32 37 37 36 30 65 ;tag=7463f27760e0100 64 35 31 37 0d 0a 43 61 6c 6c 2d 49 44 3a 20 62 d517..Call-ID: b0110 30 30 63 64 37 63 33 33 33 37 39 63 66 33 65 61 00cd7c33379cf3ea 0120 35 35 63 65 38 37 30 38 64 33 36 33 63 35 64 40 55ce8708d363c5d@0130 31 39 32 2e 31 36 38 2e 31 2e 35 30 0d 0a 43 53 192.168.1.50..CS0140 65 71 3a 20 31 39 30 33 37 39 39 38 36 31 20 53 eq: 1903799861 S 0150 55 42 53 43 52 49 42 45 0d 0a 52 6f 75 74 65 3a UBSCRIBE..Route:0160 20 3c 73 69 70 3a 31 39 32 2e 31 36 38 2e 31 2e sip:192.168.1.0170 31 3b 6c 72 3e 0d 0a 53 75 70 70 6f 72 74 65 64 1;lr..Supported 0180 3a 20 74 69 6d 65 72 0d 0a 45 78 70 69 72 65 73 : timer..Expires0190 3a 20 38 36 34 30 30 0d 0a 45 76 65 6e 74 3a 20 : 86400..Event: 01a0 6d 65 73 73 61 67 65 2d 73 75 6d 6d 61 72 79 0d message-summary. 01b0 0a 43 6f 6e 74 61 63 74 3a 20 32 30 33 20 3c 73 .Contact: 203 s01c0 69 70 3a 32 30 33 40 31 39 32 2e
Re: [asterisk-users] Mutiple Homes one asterisk box
You can place the phones at each house in a different context. Trunks, too. On 7/10/06, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote: I have a asterisk box up and running great. I have another house in mybackyard that also wants to use my asterisk box. I am running trixbox now and have two POTS lines connected to digium TDM400P as well as 1voip line for long distance. I would like to keep these two houses asseperate as possible (one POTS line for one house the other POTS for other house and share the VOIP line). What is the best way to go aboutdoing this? Both houses will have Budgetone sip phones and share thesame ethernet network. Can I install two instances of asterisk on thesame box or is there a better way? Any suggestions? ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Asterisk in Seattle
Doug, Cheer up! There's some great beer brewed in Montana! Have a Moose Drool and get down to some creative resume re-inventing. On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote: I'm in Williston, North Dakota and we have an office in Billings, MT. He'sright. We are 500 miles form civilization! :) -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of DouglasGarstangSent: Wednesday, July 05, 2006 10:00 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion;asterisk-users@lists.digium.com Subject: RE: [asterisk-users] RE: Asterisk in SeattleIt can't be that bad there in Anchorage. I'm in Billings, MT, which is abouthalf the size of Anchorage, and sometimes (no, wait... most times) it seems like I'm 500 miles from civilsation. Wait, I AM 500 miles from civilisation! -Original Message- From: Josh Reineke [mailto:[EMAIL PROTECTED]] Sent: Wed 7/5/2006 8:03 PM To: asterisk-users@lists.digium.com Cc: Subject: [asterisk-users] RE: Asterisk in Seattle I work for a medium size business in Anchorage, AK running two installations with about 30 handsets a piece.They've both been in service for a couple of years. I'm in Seattle fairly frequently, being it's the metropolis closestto Anchorage.I'd be jazzed if there was a user group there and would be willing to help in it's formation. Josh Message: 15 Date: Wed, 5 Jul 2006 14:00:35 -0600 From: Douglas Garstang [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk in Seattle To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID:[EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 All, Anyone know of any companies (small, large) that are using, experimenting with, deploying, and so on, Asterisk in Washington state, most likely in and around Seattle? I'm curious from an employment perspective. :) Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
And I thought these were for your personal use. As things stand today with the message waiting lamp problems, I would not want to offer these phones to a customer using SIP. They work great as h.323 phones connected to an Avaya system, or SIP connected to an Avaya system. Using them with Open Standards systems, they're still a bit wonky. On 7/4/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi, You can call me by my first name (Silviu) :)) I have made the changes to the settings file, I have removed the LDAP-related settings - nothing changes... The file is still taken into account, as other changes affect the phone, but the SIP fields stay desperately blank... I don't think I'll wait for the next firmware release, I'm currently evaluating several Siemens optiPoint phones (SIP) which look good so far. I have to get things moving, the customer won't wait forever for the Avaya phones to work.c However I'm a bit disappointed to leave things as they are, I have a feeling of ... failure? I guess I'll still try some thing or another in my (inexistent) spare time. Thanks for your help, Silviu From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Tom Lynn Sent: 04 July 2006 03:57 To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Herchi,I want you to re-read my last e-mail very carefully. Your response does not mention at all my guess that the three SP_DIRSRVR variables may be giving you trouble. I'm still interested in knowing what happens if you remove them from your settings file. Also, I have heard a rumour that there will be a new firmware release on July 10th. Actually, I just clicked the feedback button on their web page for the firmware download and asked. They responded on the first business day (unusual for Avaya), indicating 7/10 is the approximate release date. So there you have my source Let me know On 7/3/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi, I had edited out all lines starting with a #, which is ot right, as the marker for comments is##... See below for the entire file. I just tried the configuration throughDHCP, by setting the 176 option to point to the right TFTP server and also to the right SIP proxy. The Avaya boot test application is not complaining, but the phones ... do I need to say it? *sigh* SET DOMAIN company.com SET DNSSRVR 204.140.111.43SET PHNCC 352 SET PHNDPLENGTH 4SET PHNIC 00SET PHNOL 0SET SYSLANG EnglishSET APPSTAT 1SET RESTORESTAT 1SET AGCHAND 0SET AGCHEAD 0 SET AGCSPKR 0SET SNTPSRVR 204.140.111.200SET DSTOFFSET 1SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2LSET GMTOFFSET -5:00SET DATESEPARATOR /SET DATETIMEFORMAT 3 SET SIPDOMAIN slt05.company.agn SET SIPPROXYSRVR 204.140.111.219SET SIPPORT 5070 SET SIPREGISTRAR 204.140.111.219 SET DIALPLAN [234]xxx|55SET DIALWAIT 3SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SP_DIRSRVR 10.1.1.1SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya.comIF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610IF $MODEL4 SEQ 4620 goto SETTINGS4620IF $MODEL4 SEQ 4621 goto SETTINGS4621IF $MODEL4 SEQ 4622 goto SETTINGS4622IF $MODEL4 SEQ 4625 goto SETTINGS4625IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END # SETTINGS4602goto END# SETTINGS4610 SET WMLHOME http://support. avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.246 SET WMLPORT 3128goto END # SETTINGS4620goto END# SETTINGS4621goto END# SETTINGS4622goto END# SETTINGS4625goto END# SETTINGS4630 SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htmSET PHNEMERGNUM 112goto END # END From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom Lynn Sent: 01 July 2006 18:18 To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Is the text shown below the ENTIRE file? It looks like all of the settings for the individial phone models are missing. I'm not sure what the consequences of branching to the 4610 section will be if it doesn't exist. Also, I don't use the SP_DIRSRVR values. What happens if those three entries are removed? SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com I can't find these three entries anywhere in my 46xx settings file.I also cannot find them in the lan admin guide from the manufacturer.They seem to be somewhat like the ldap options for the 4630 phone, but those didn't have a leading SP_ prefix on the variable name. Why don't you comment them out and see what happens?Tom Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR
Re: [asterisk-users] Asterisk in Seattle
I don't know of anybody using it inbusiness, but I'm curious to find out if there are any user groups formed or forming in the Seattle Area. On 7/5/06, Douglas Garstang [EMAIL PROTECTED] wrote: All,Anyone know of any companies (small, large) that are using, experimenting with, deploying, and so on, Asterisk in Washington state, most likely in and around Seattle? I'm curious from an employment perspective. :) Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Asterisk in Seattle
Doug,Two Points- User groups are excellent places to network and make contacts- Asterisk skills will translate into other telecom or non telecom related fields. I work for a fortune 500 company and I'm responsible for a bunch of Avaya systems. None of the people on my team had Avaya experience before they joined, but they were skilled in other telecom areas. I'm sure Seattle can support one more good telecom engineer, even if it's a short term gig while you look for your dream * job. If I hear of anything, I'll post it here. Good luck.On 7/5/06, Douglas Garstang [EMAIL PROTECTED] wrote: It can't be that bad there in Anchorage. I'm in Billings, MT, which is about half the size of Anchorage, and sometimes (no, wait... most times) it seems like I'm 500 miles from civilsation. Wait, I AM 500 miles from civilisation! -Original Message-From: Josh Reineke [mailto:[EMAIL PROTECTED]]Sent: Wed 7/5/2006 8:03 PMTo: asterisk-users@lists.digium.comCc:Subject: [asterisk-users] RE: Asterisk in SeattleI work for a medium size business in Anchorage, AK running twoinstallations with about 30 handsets a piece.They've both been in service for a couple of years.I'm in Seattle fairly frequently, being it's the metropolis closest toAnchorage.I'd be jazzed if there was a user group there and would bewilling to help in it's formation. JoshMessage: 15Date: Wed, 5 Jul 2006 14:00:35 -0600From: Douglas Garstang [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk in SeattleTo: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Message-ID:[EMAIL PROTECTED]Content-Type: text/plain; charset=iso-8859-1 All,Anyone know of any companies (small, large) that are using,experimenting with, deploying, and so on, Asterisk in Washington state,most likely in and around Seattle? I'm curious from an employment perspective. :)Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
Herchi,I want you to re-read my last e-mail very carefully. Your response does not mention at all my guess that the three SP_DIRSRVR variables may be giving you trouble. I'm still interested in knowing what happens if you remove them from your settings file. Also, I have heard a rumour that there will be a new firmware release on July 10th. Actually, I just clicked the feedback button on their web page for the firmware download and asked. They responded on the first business day (unusual for Avaya), indicating 7/10 is the approximate release date. So there you have my source Let me know On 7/3/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi, I had edited out all lines starting with a #, which is ot right, as the marker for comments is##... See below for the entire file. I just tried the configuration throughDHCP, by setting the 176 option to point to the right TFTP server and also to the right SIP proxy. The Avaya boot test application is not complaining, but the phones ... do I need to say it? *sigh* SET DOMAIN company.com SET DNSSRVR 204.140.111.43SET PHNCC 352SET PHNDPLENGTH 4SET PHNIC 00SET PHNOL 0SET SYSLANG EnglishSET APPSTAT 1SET RESTORESTAT 1SET AGCHAND 0SET AGCHEAD 0SET AGCSPKR 0SET SNTPSRVR 204.140.111.200SET DSTOFFSET 1SET DSTSTART 1SunApr2LSET DSTSTOP LSunOct2LSET GMTOFFSET -5:00SET DATESEPARATOR /SET DATETIMEFORMAT 3SET SIPDOMAIN slt05.company.agnSET SIPPROXYSRVR 204.140.111.219SET SIPPORT 5070 SET SIPREGISTRAR 204.140.111.219 SET DIALPLAN [234]xxx|55SET DIALWAIT 3SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3SET REGISTERWAIT 120SET SP_DIRSRVR 10.1.1.1SET SP_DIRSRVRPORT 389SET SP_DIRTOPDN ou=People,o=avaya.comIF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610IF $MODEL4 SEQ 4620 goto SETTINGS4620IF $MODEL4 SEQ 4621 goto SETTINGS4621IF $MODEL4 SEQ 4622 goto SETTINGS4622IF $MODEL4 SEQ 4625 goto SETTINGS4625IF $MODEL4 SEQ 4630 goto SETTINGS4630goto END# SETTINGS4602goto END# SETTINGS4610SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wmlSET WMLPROXY 204.140.111.246 SET WMLPORT 3128goto END# SETTINGS4620goto END# SETTINGS4621goto END# SETTINGS4622goto END# SETTINGS4625goto END# SETTINGS4630SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htmSET PHNEMERGNUM 112goto END# END From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 01 July 2006 18:18To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Is the text shown below the ENTIRE file? It looks like all of the settings for the individial phone models are missing. I'm not sure what the consequences of branching to the 4610 section will be if it doesn't exist. Also, I don't use the SP_DIRSRVR values. What happens if those three entries are removed? SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com I can't find these three entries anywhere in my 46xx settings file.I also cannot find them in the lan admin guide from the manufacturer.They seem to be somewhat like the ldap options for the 4630 phone, but those didn't have a leading SP_ prefix on the variable name. Why don't you comment them out and see what happens?Tom Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR / SET DATETIMEFORMAT 3 SET DIALPLAN [234]xxx|55 SET DIALWAIT 3 SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SIPDOMAIN sip.mycompany.com SET SIPPROXYSRVR 204.140.111.219 SET SIPPORT 5070 (this is not a typo) SET SIPREGISTRAR 204.140.111.219 SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
Is the text shown below the ENTIRE file? It looks like all of the settings for the individial phone models are missing. I'm not sure what the consequences of branching to the 4610 section will be if it doesn't exist. Also, I don't use the SP_DIRSRVR values. What happens if those three entries are removed? SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com I can't find these three entries anywhere in my 46xx settings file.I also cannot find them in the lan admin guide from the manufacturer.They seem to be somewhat like the ldap options for the 4630 phone, but those didn't have a leading SP_ prefix on the variable name. Why don't you comment them out and see what happens?Tom Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR / SET DATETIMEFORMAT 3 SET DIALPLAN [234]xxx|55 SET DIALWAIT 3 SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SIPDOMAIN sip.mycompany.com SET SIPPROXYSRVR 204.140.111.219 SET SIPPORT 5070(this is not a typo) SET SIPREGISTRAR 204.140.111.219 SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
I'm setting both values like you are: SET SIPREGISTRAR xxx.xxx.xxx.xxx SET SIPPROXYSRVR xxx.xxx.xxx.xxx I don't notice a difference in how these settings appear in our respective 46xxsettings.txt files. On 6/29/06, Henk [EMAIL PROTECTED] wrote: Did you try to manually to change the parameters of the phone? When you power the phone up then are you able to enter manually the parameter when you hit *. I am using a 4610 with Release 2.2 but I am not using the capability to upload the settings from the server. Henk From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Herchi SilviuSent: donderdag 29 juni 2006 15:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Avaya 4610sw SIP setup problem I just tried serving the files off Apache, port 80, no change... Most parameters are taken into account by the phone, except for SIP proxy and SIP registrar... Coud someone post an excerpt from their 46xxsettings.txt where I could see the format they use? Thank you in advance, Silviu From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 29 June 2006 00:33 To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since. On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2 ). Thanks, Silviu From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 28 June 2006 05:35 To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem… I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying Registering… for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty… I have tried specifying them with and without quotes, by hostname, by IP address, … Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR / SET DATETIMEFORMAT 3 SET DIALPLAN [234]xxx|55 SET DIALWAIT 3 SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SIPDOMAIN sip.mycompany.com SET SIPPROXYSRVR 204.140.111.219 SET SIPPORT 5070(this is not a typo) SET SIPREGISTRAR 204.140.111.219 SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since. On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem… I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying Registering… for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty… I have tried specifying them with and without quotes, by hostname, by IP address, … Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR / SET DATETIMEFORMAT 3 SET DIALPLAN [234]xxx|55 SET DIALWAIT 3 SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SIPDOMAIN sip.mycompany.com SET SIPPROXYSRVR 204.140.111.219 SET SIPPORT 5070 (this is not a typo) SET SIPREGISTRAR 204.140.111.219 SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
Which version of firmware are you using?On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem… I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying Registering… for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty… I have tried specifying them with and without quotes, by hostname, by IP address, … Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR / SET DATETIMEFORMAT 3 SET DIALPLAN [234]xxx|55 SET DIALWAIT 3 SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SIPDOMAIN sip.mycompany.com SET SIPPROXYSRVR 204.140.111.219 SET SIPPORT 5070(this is not a typo) SET SIPREGISTRAR 204.140.111.219 SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP
The avaya softphone (an entitlement in recent versions of the PBX) will dial from outlook and allows clicking of phone numbers from within web based content. I'm not sure it works from other office apps, though. On 6/27/06, Rodney G. McDuff [EMAIL PROTECTED] wrote: Brian Capouch wrote: It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot.The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.htmlIts all bad news interoperability wise.--Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group|Ex luce ad tenebrasInformation Technology Services|The University of Queensland |EMAIL: [EMAIL PROTECTED] |TELEPHONE: +61 7 3365 8220 |-- Forwarded message --From: Christian Schlatter [EMAIL PROTECTED]To: 'tf-vvc' [EMAIL PROTECTED]Date: Tue, 27 Jun 2006 09:38:17 -0400Subject: Re: [tf-vvc] Microsoft sends message on 'unified messaging'Cătălin Meiroşu wrote: fyi (apologies to those that have already read about it) Microsoft sends message on 'unified messaging' The software maker on Monday announced its vision for so-called unified messaging, which brings together e-mail, instant messaging, telephony and Web conferencing. It also introduced a series of products coming over the next year that should help achieve this. The goal is to free workers from having to guess which mode is the best to use to reach co-workers and others. [...]Although MS claims for SIP standards compliance, their UM products arestill using proprietary protocols. I can't register my x-lite with anLCS or use the MS communicator together with my SER proxy. So it's all or nothing, either you're an MS shop from A to Z or you can't really usetheir UM offerings.Tom Keating has written a nice article about that: http://blog.tmcnet.com/blog/tom-keating/microsoft/microsoft-office-communications-server-2007.aspChristian___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings
Those are the only files that come to mind. On 6/23/06, Erick Perez [EMAIL PROTECTED] wrote: Tom, just to make sure im on the right track.What files do you tweak?sip.conf, the ones from avaya and anything else?On 6/22/06, Tom Lynn [EMAIL PROTECTED] wrote: Nope.Let me know if you do.I've suspended my efforts until I see a new version of firmware available on the Avaya web site. On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote: Thanks for your comments Tom. Indeed the MWI and the programmable buttons are the only things that do not work for me. Besides that, the phone is great and the audio quality is superb. Did you managed somehow to make the MWI work? Will keep searching the net, the 4602 page is somehow poor on the documentation. On 6/21/06, Tom Lynn [EMAIL PROTECTED] wrote: Well, I wouldn't say nobody.I do and I've corresponded with a few people that do. There's a page on voip-info.org dedicated to the Avaya 4602 telephone and SIP (I'm hoping I'm not the only reader of that page). When I've used my Avaya phone in conference (FWD CoffeeHouse), I've had people sincerely compliment me on the quality of sound with my phone. But.. Avaya has a few things working against it within the context of Asterisk: * MWI just doesn't work (If you insist on trying it, get ready for your phone to lose it's registration with * every hour or so) * Dial strings beginning with * character appear to go nowhere with these phones * They're perceived as rather expen$ive * As a company, they're simply not focused on * since it doesn't help sell any of their other product.They prefer selling things that drive maintenance contract revenue and, let's face it, the phone is the commodity appliance that connects to *.Even within the enterprise space, very few carry maintenance on their telephone sets anymore. Funny anectdote:Avaya loves showing Cisco 79xx phones with a SIP load registered to their PBX systems with a Powered By Avaya background. They claim that, unlike Cisco, they will accept third party SIP clients registering to their system.However, they really don't provide any kind of support for their phones used with a system other than their own.My Mom used to call that the Pot calling the Kettle Black. Good phone, great sound, just no support and a bit wonky on the features. My 2 cents. On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote: nobody uses avaya phones with asterisk? On 6/20/06, Erick Perez [EMAIL PROTECTED] wrote:Hi, I setup my tftp to send SIP configurations (the bin files) to the avaya phone. When it finished loading and rebooting it asked for theextension and the password and the asterisk ip address. I had to inputthat manually and is now working perfectly with asterisk. what is the format of the text files to make this phone load theasterisk ip, extension number, codec used, password as well as toconfigure message waiting indicator and maybe modify some of the buttons (such as just pressing one of the available programmable buttons to access voicemail). I have 10 more of these phones and iwant to do provisioning automatically. in the 46xxsettings.txt file there are no such parameters thanks, -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de PanamaCel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth
Re: [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings
Nope. Let me know if you do. I've suspended my efforts until I see a new version of firmware available on the Avaya web site.
On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote:
Thanks for your comments Tom. Indeed the MWI and the programmablebuttons are the only things that do not work for me. Besides that, the
phone is great and the audio quality is superb.Did you managed somehow to make the MWI work?Will keep searching the net, the 4602 page is somehow poor on the documentation.On 6/21/06, Tom Lynn
[EMAIL PROTECTED] wrote: Well, I wouldn't say nobody.I do and I've corresponded with a few people that do. There's a page on voip-info.org
dedicated to the Avaya 4602 telephone and SIP (I'm hoping I'm not the only reader of that page).When I've used my Avaya phone in conference (FWD CoffeeHouse), I've had people sincerely compliment me on the quality of sound with my phone.
But.. Avaya has a few things working against it within the context of Asterisk: * MWI just doesn't work (If you insist on trying it, get ready for your phone to lose it's registration with * every hour or so)
* Dial strings beginning with * character appear to go nowhere with these phones * They're perceived as rather expen$ive * As a company, they're simply not focused on * since it doesn't help sell
any of their other product.They prefer selling things that drive maintenance contract revenue and, let's face it, the phone is the commodity appliance that connects to *.Even within the enterprise space, very few
carry maintenance on their telephone sets anymore. Funny anectdote:Avaya loves showing Cisco 79xx phones with a SIP load registered to their PBX systems with a Powered By Avaya background.They
claim that, unlike Cisco, they will accept third party SIP clients registering to their system.However, they really don't provide any kind of support for their phones used with a system other than their own.My Mom
used to call that the Pot calling the Kettle Black. Good phone, great sound, just no support and a bit wonky on the features. My 2 cents.
On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote: nobody uses avaya phones with asterisk? On 6/20/06, Erick Perez
[EMAIL PROTECTED] wrote: Hi, I setup my tftp to send SIP configurations (the bin files) to the avaya phone. When it finished loading and rebooting it asked for the extension and the password and the asterisk ip address. I had to input
that manually and is now working perfectly with asterisk. what is the format of the text files to make this phone load the asterisk ip, extension number, codec used, password as well as to
configure message waiting indicator and maybe modify some of the buttons (such as just pressing one of the available programmable buttons to access voicemail). I have 10 more of these phones and i
want to do provisioning automatically. in the 46xxsettings.txt file there are no such parametersthanks,--
Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama
Cel Panama. +(507) 6694-4780 --
Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780
___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
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http://lists.digium.com/mailman/listinfo/asterisk-users--Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de PanamaCel Panama. +(507) 6694-4780___--Bandwidth and Colocation provided by
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings
Well, I wouldn't say nobody. I do and I've corresponded with a few people that do. There's a page on voip-info.org dedicated to the Avaya 4602 telephone and SIP (I'm hoping I'm not the only reader of that page). When I've used my Avaya phone in conference (FWD CoffeeHouse), I've had people sincerely compliment me on the quality of sound with my phone. But..Avaya has a few things working against it within the context of Asterisk:* MWI just doesn't work (If you insist on trying it, get ready for your phone to lose it's registration with * every hour or so) * Dial strings beginning with * character appear to go nowhere with these phones* They're perceived as rather expen$ive* As a company, they're simply not focused on * since it doesn't help sell any of their other product. They prefer selling things that drive maintenance contract revenue and, let's face it, the phone is the commodity appliance that connects to *. Even within the enterprise space, very few carry maintenance on their telephone sets anymore. Funny anectdote: Avaya loves showing Cisco 79xx phones with a SIP load registered to their PBX systems with a Powered By Avaya background. They claim that, unlike Cisco, they will accept third party SIP clients registering to their system. However, they really don't provide any kind of support for their phones used with a system other than their own. My Mom used to call that the Pot calling the Kettle Black. Good phone, great sound, just no support and a bit wonky on the features.My 2 cents.On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote:nobody uses avaya phones with asterisk?On 6/20/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, I setup my tftp to send SIP configurations (the bin files) to the avaya phone. When it finished loading and rebooting it asked for the extension and the password and the asterisk ip address. I had to input that manually and is now working perfectly with asterisk. what is the format of the text files to make this phone load the asterisk ip, extension number, codec used, password as well as to configure message waiting indicator and maybe modify some of the buttons (such as just pressing one of the available programmable buttons to access voicemail). I have 10 more of these phones and i want to do provisioning automatically. in the 46xxsettings.txt file there are no such parameters thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 --Erick PerezPanama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Copper or T1 Fiber Line
HDSL can sometimes deliver service where copper pairs are nearly exhausted. In other words, if you're down to your last pair of copper, a normal two-pair T1 cannot be delivered, whereas T1 via HDSL can. On 6/17/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] a écrit : Thanks for the inso... So T1 lines in the United States also use copper lines from the company to the telephone exchange in some installations? What's the benefit to the subscriber to this?I don't think there is any difference. The E1s I've got at home arebrought with copper HD-SDSL and they work just fine.Cheers, Jean-Michel.--Jean-Michel Hiver - http://ykoz.net/Découvrez la Réunion des Technologies IP TelecomTEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] free sun boxes
Whare are they located?On 6/17/06, Bob Knight [EMAIL PROTECTED] wrote: I have 4 sparc based sun boxes I am about to pay money so I canget rid of them.They are running older versions of Solaris.You should be able to load Solaris 10 and play around with *on them.Time to clean the office: 3 Ultra 51 Sparcstation 5I also have a box full of Sun keyboards and mice.Contact me offline if you want them.I've had many good years of development on them and it killsme to just toss them, but the office is just too damn cluttered. thanks, bk...--Bob Knight[-w] the work option[EMAIL PROTECTED]925-449-9163___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can this config sustain 30 users?
Don't forget to be sure your power supplies are reliable, and if necessary redundant. On 6/13/06, Colin Anderson [EMAIL PROTECTED] wrote: Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2PRI's and we regularly have 40-60 channels up, no problem (believe me, if there was a problem I'd have 200 guys freaking on my head). I rarely see 30% single-CPU usage, and that's only when Sendmail is invoked to send out avoicemail.But yes, transcoding and reasonable echocancel values is key. If you are connecting to the PSTN, ulaw all the way. If you are connecting to aprovider, use the codec of your choice as long as your provider supports it,and make sure every phone and endpoint is set to use the same codec. I also have 30 IAX remote sites that support from 1 to 5 users, on P-II233's. I use them because they are bulletproof and they are so cheap ifsomething gets hosed we just throw it away and put in another one. Again, no problemMaybe try your cheapo machine and if it doesn't work try a better box. Youalready have the cheap machine, and the card will remain the same regardlessof what box you use.-Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 13, 2006 9:15 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can this config sustain 30 users? Well thanks all for your responses. My original intention was toaddress the mistic know-how about machine calculations, and I stillfeel the shadows remain.Why? Because to achieve a 24 user PBX-only/One E1, I was going to install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1with two sata3 disks.Now This thread tells me that my dual core pentium d (a 700$ computer)will do the work. (the other equipment costs about 3500.00$). I dorealize that i must minimize transcoding (ulaw all the way) but you'retelling me it will work for 24 users (let's say 30 for round numbers)all with SIP phones in an IP network.Below are some comments that i found googling and doing some calculations myself. I do not enforce or deny any of them, please feelfree to tell me if Im wrong.(not confirmed)a- A voice channel takes 30mhz (60mhz in duplex mode).So A 2.66 Ghz CPU can sustain about 43 calls (2600mhz/60mhz=43calls), not taking into account other factors that may increase/decrease thenumber of calls at the same time.b- 24 users talking ulaw (-+ 80kbps per channel) consume 1920kbps andin full duplex they consume 3840kbps (about 3.75 megabits/s).c- To Calculate the bandwidth DDR memory can achieve (example PC4200),to get the transfer rate, multiply the width of the module (8 Bytes)by the rated speed of the memory module (in MHz): (8 Bytes) x (533 MHz/second) = 4,264 Mbytes/second or 4.2 Gbytes/second (34gigabits/s),hence the name PC4200So, will all of this in mind,CPU Dual Core 533FSB, 2.66 Ghz speedDDR533mhz, One gigabyte. (2x512)Two Sata disks (each sata pumps 1.5 gigabits/s)Motherboard Intel 945 at 533FSBMeans that the cpu,the ram and the board can achieve (see point b)about 34 gigabits of data transfer, but 24 users only generate 3.75megabits. So this is more than covered. However if we take into account the lowest performing component onthis system (the sata disks) we go down to 1.5gbits/s which stillseems to be enough.Please please correct me if im wrong (or crazy) Thanks,References:http://en.wikipedia.org/wiki/Front_side_bus (bus bandwidth table) http://www.acme.com/build_a_pc/bandwidth.htmlhttp://www.lostcircuits.com/memory/ddrii/ http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_busOn 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote: Erick Perez wrote: I just don't want to install it and then after a 5th user going to call someone the asterisk begin to crash due to lack of resuources. Check the wiki for SIP load generation tools you can use to test your setup on any number of calls you like. ___ --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--Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de PanamaCel Panama. +(507) 6694-4780___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update
[Asterisk-Users] What's the current state of using shared lines in asterisk?
I'm trying to get to where I can program a phone to have 3-6 buttons each representing the same extension number. Also, I'd like to have them appear on more than one phone like key systems do.Is asterisk able to set up shared lines in this manner yet? Tom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help Avaya 4606
The 4606 is a h.323 based phone. There is no SIP image to use with this phone. On Fri, 12 May 2006 11:11:48 -0500, you wrote: Hello all, I have asterisk working well with, Sipura, but I do not manage to form several phones avaya 4606, someone could have formed one avaya with asterisk? is it possible? update the firmware of the phone, but I do not achieve that it registers, I hope that someone could help me greetings to all Carlos Rojas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mixmonitor
On Tue, 15 Nov 2005 11:51:33 -0500, you wrote: On 11/15/05, Brian Roy [EMAIL PROTECTED] wrote: On 11/14/05, BJ Weschke [EMAIL PROTECTED] wrote: There is a known issue right now where using mixmonitor with chan_local is going to cause an unintentional disconnect. Are you using Local/ with this setup? BJ, Thanks for the response. No, I've got nothing going though chan/local at all. It's a real straigh-forward zap to sip bridge. Nothing fancy. I'm going to try and route my calls over to another box via iax today and see if that makes any difference. The mixmonitor will be looking at sip to iax then. Let me know if you think I should file a bug on this. -Brian I think that you should. There's no known issues that I'm aware of with the configuration you're speaking of. Thanks BJ I'm having problems with this as well. It's a couple of months later and I'm wondering if anybody has gotten to the bottom of this. I am calling from a sip phone to an IAX softphone. The calls don't go much longer than a minute before cutting off. Thanks - Tom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording queue calls
Faris,
Is there a way to have * send save these in an off-server location? Or
have * e-mail them via smtp and then delete them from the server
automatically?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: Saturday, December 24, 2005 10:18 AM
To: Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] recording queue calls
Dov Bigio wrote:
Hi,
When I set monitor-format=wav49 on file queues.conf for a queue,
Asterisk records calls at /var/spool/asterisk/monitor. But the file
names it users are the call-ids of the calls.
Is there a way to change that, and use information such as date, time,
agent and queue to build the filename?
It would make the localization of such files much more easy.
Other useful that I miss is the capability to to allow the files to be
stored in different directories, such as
/var/spool/asterisk/monitor/queue1,
/var/spool/asterisk/monitor/queue2,
and so on, based on the queuename. Is this possible by any means?
Hi,
Yes. All you need to do is use the following in your extension.conf at
the point before you call the queue
SetVar(MONITOR_FILENAME=foo)
or, if you are using 1.2.x
Set(MONITOR_FILENAME=foo)
For example, I have:
Set(MONITOR_FILENAME=${TIMESTAMP}-${CALLERIDNUM}-to-SALES-${UNIQUEID})
and then a little later on:
Queue(salesqueue|t|||60)
in my extensions.conf
Which sets the monitor filename to start with a timestamp, then the CID
of the caller, then the to-SALES is what I use to differentiate
between queues (I'd have a different Set command for a different queue).
I then add the UNIQUEID as a just in case to make absolutely sure
there's no way I'd ever have two files of the same name.
I hope this helps,
Faris.
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RE: [Asterisk-Users] recording queue calls
Rsync could happen overnight, but I'm really looking for a solution that removes the recording from the system so as not to kill my limited storage. I'll be running astlinux from a 256mg Compact Flash card and 256meg of USB keydisk space for configs and recordings. I need to move 'em off fast. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: Saturday, December 24, 2005 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] recording queue calls Tom Lynn wrote: Faris, Is there a way to have * send save these in an off-server location? Or have * e-mail them via smtp and then delete them from the server automatically? I'm sure there is a very technical way of doing it. For example if I remember correctly you can set your own script to run to join the two sides of an audio recording (something I tried using to solve the problem I'm having with joining two sides of a conversation, but with no luck). You could add a mail command to the script to do what you want. I'm afraid I don't remember the exact details of how this is done, but I think I came across it when searching for asterisk call recording on Google. There was a full script for an alternative mixing solution. Or you could use rsync, running every hour or every day as a cron job, to synchronise the /var/spool/asterisk/monitor directory on the machine tasking the calls with a second server. e.g. rsync -e ssh -avz /var/spool/asterisk/monitor/ [EMAIL PROTECTED]:~/monitorbackup You'd need to set up a passwordless private/public key combination for this to work automatically though. There may also be issues with the rsync job using too much bandwidth and causing audio quality problems. Hmm... Well, I'm sure someone who know more than me on this topic will pipe up on this! Faris. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.7/214 - Release Date: 12/23/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.7/214 - Release Date: 12/23/2005 ___ --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] Anybody having trouble terminating calls at Voxee? eom
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Re: [Asterisk-Users] Goi ax.com DID not working anymore?
I've always received a busy signal when I dial mine. Additionally, I can't see any server messages to indicate that goiax is even attempting to call my system, although I continue to trouble-shoot. I have a 413-230- number. Thanks On Thu, 20 Oct 2005 15:32:08 -0600, you wrote: I've been using my goiax.com http://goiax.com DID for a few days now and it is no longer working. I get the number or code you dialed can not be found. I haven't touched any configs or anything on the asterisk box since it was working last night. Anyone else having problems using the DID from goiax? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone Know if tha avaya 4621 IP phone work wiht asteisk?
Seems to, mostly, but not all the time.. There have been some previous posts about this phone. I'm waiting for the next firmware release before testing further. Mute kills audio in both directions. Mine doesn't seem to want to dial any numbers that have * in them like *98. My Sipura 3000 returns busy only when the 4621 calls it. Rings and answers to my softphones. After a few minutes, it won't dial other numbers. Returns fast busy upon dialing. * shows it as unreachable. I keep seeing messages on the * console saying the phone received an invalid subscription. And I quote: -- Got SIP response 400 Invalid Subscription-State back from xxx.xxx.xxx.xxx where the x'd out address is the address of the Avaya 4621 set. If anybody has any useful advice on troubleshooting this, I'd appreciate it. Tom On Fri, 14 Oct 2005 13:49:18 -0400, you wrote: Does anyone Know if tha avaya 4621 IP phone work wiht asteisk? if it work it has featuras working Thanks Ignacio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya 4620/4640 SIP firmware
On Sun, 9 Oct 2005 14:28:14 -0600, you wrote: The initial release of Avaya's SIP firmware for the 4620 phone was released on August 17, 2005. It is available from support.avaya.com. That said, I have had mixed results with this phone. I'm sure it works with Avaya gear, but it's glitchy with *. For instance, I noticed audio stops in BOTH directions when I use the mute button. After a period of time, the phone doesn't really un-register, but nor will it dial. I'd love to hear how other people deal with these problems. I've tested for over a week and I'm ready to wait for the next firmware release before proceeding further. Tom Does anybody know if Avaya has a test SIP firmware available for 4620 and 4640 IP phones? The 46xx SIP image from their website is a combo download with SIP for the 4602, and h323 for the the 4620 and 4640. It looks like they demo'd a SIP image for the 4640 as far back as 2004: http://www.sip.org/von/2004/boston/slides/DSC_0042.php Thanks, Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM tts engine integration
On Sun, 02 Oct 2005 00:53:03 -0700, you wrote: I wrote a very very simple shell script and an even simplier macro to use the IBM TTS engine within asterisk for prompts. While its free you are limited on the number of requests you can do within a day. If anyone is interested its available at http://www.0xdecafbad.com/Asterisk-Text-to-Speech.html Nice solution, but what will you do if/when IBM pulls their demonstration page? Hopefully, by then you will have cached all of the necessary recordings. ___ --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] Avaya 4620 hardphone
I've just interfaced an Avaya 4621 set to [EMAIL PROTECTED] It's running the 2.2 SIP firmware released August 17th. I've run into some strange behavior with this phone. 1st, in order to get two way audio, I had to tell * that it was behind a NAT even though it is on the same subnet as the * server. Second, it doesn't seem to be able to dial any extensions or features that use the * digit in them. The phone remains off hook, and I get a fast busy after about thirty seconds. I also can't find any log entries that shoe the attempt was even seen by the server. I've completely disabled the internal dial plan inside the phone and the problem persists. Third, the mute button seems to disable audio in both directions, not simply the local microphone. Has anybody else interfaced one of these phones to *? If so, did you experience these same problems and did you find any solutions? I believe this is the first public release of SIP firmware for this phone and I know it hasn't been out that long. Thanks - Tom ___ --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
