[Asterisk-Users] Video Conferencing
Can the asterisk system support video conferencing? ___ --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 man in the middle
I am planing on doing it a daemon that can live on the asterisk box or any box that can run unix and iptables. I will need to reroute packets aimed for providers box to the box where the daemon lives. In my case using a low power(15watts) is the way to go. If your asterisk box has the spare power to run the daemon and iptables that is fine. In my case as a home user I am planning on moving from PC based asterisk to low power box (linksys WRT54g) which should be able to handle 2 active connection with about 8 extension. On 12/31/05, Stewart Nelson [EMAIL PROTECTED] wrote: Hi Mike, This is wanted because using to ATA back to back creates a number of problems with echo. Also a delay for CID and problems with DTMF decoding. Keep everything digital is the way to go. Agreed.But before getting started with Asterisk, I posted a similar ideato the group; it was met with a quite cool reception, on and off-list.See http://lists.digium.com/pipermail/asterisk-users/2004-October/068932.html .I ended up avoiding Vonage and using multiple other providers.That said, I believe that many users of non-BYOD ITSPs would benefit from a proxy such as you describe.Unfortunately, I'm not aware of anyonethat has implemented it yet.If you undertake such a project, IMO youshould do it in Asterisk, or as a separate process that can run on the same machine as Asterisk, because many more people would use it andcontribute to its development.--Stewart___--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] voicemail/privacy system
Original Message From: Eck [EMAIL PROTECTED] To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Sent: Saturday, December 31, 2005 8:26 PM Subject: RE: [Asterisk-Users] voicemail/privacy system If you dont want to get too stuck into the guts of Asterisk yet, the [EMAIL PROTECTED] distribution can do all you have requested with a one button install web configuration via AMP. Personally I think its a great place to start with asterisk whatever your requirements as it makes a good base without having to go through the drudgery of installing asterisk the requirements/add-ons piecemeal, espically AMP, as the prereqs are a stress! (mumbles something about a, thankfully forgotten, nightmarish FreeBSD Asterisk/AMP install then fades into background, wimpering) :) I installed asterisk from the FreeBSD ports, but then took the config's from the live asterisk cd. That was fairly painless. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Original Message From: Peter Bowyer [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 31, 2005 11:34 AM Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP Hi all Slightly OT but I know a lot of GS experts hang out here - I just upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk (which so far works as expected), but as a side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ? (Hmmm.. just regressed to 1.0.1.12 and it's still not working - curiouser and curiouser said Alice...) My GS BT101 have also developed problems with sync'ing to my ntp-server. I can see, using tcpdump, that the phone asks my server and gets an answer, but the display is not updated. It used to work, but now it usually doesn't, but strangely, sometime it does... Leif ___ --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] Video Conferencing
Well, the documentation states that Video Conferencing is possible. I've tried working with EyeBeam, which yielded nice Results, but anything beyond that - I can't comment. Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Sunday, January 01, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video Conferencing Can the asterisk system support video conferencing? ___ --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] Video Conferencing
Well, the documentation states that Video Conferencing is possible. I've tried working with EyeBeam, which yielded nice Results, but anything beyond that - I can't comment. Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Sunday, January 01, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video Conferencing Can the asterisk system support video conferencing? ___ --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] Recommendations on web interface for IT staff
I am proposing an Asterisk system of many servers to service multiple departments in a number of locatations to a large client. They have an IT department but their Linux skills are weak and they are likely to face a high churn rate in staff so it would not be wise to expect a high level of Linux expertise to be maintained. I am thinking it would be best to do the nitty gritty glue work at the config file level myself but have a web based interface to common tasks such as managing extensions, adding trucks, voicemail etc. They are anxious for obvious reasons that they are able to manage the system without having to call me every time they need changes. As it will be a multi-server system there will be some fairly detailed configs to put together, so I would think a [EMAIL PROTECTED] installation would not be suitable, but I haven't tested that theory so I am not against trying it. What recommendations for web management can you make from experience of larger systems? It doesn't have to be limited to free systems. I am also interested in opinions on whether you would implement one monster server to do everything and have parts to maintain it, or would your preference be to have one server per department and interlink them, keeping the hardware the same and having a standby system ready to fill in for failed systems. On one hand there is only one server to monitor, on the other there is redundancy but also complexity. I can see advantages in both approaches. -- Chris Mason ___ --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] CrystalFontz LCD display
I saw a brief discussion via Google about developing support for LCD displays, ones that you integrate into a drive bay or whatnot for server information output. Any development on this? I couldn't find much. --Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check Queue Statistics
On Sat, 31 Dec 2005 20:23:09 +0100, BJ Weschke [EMAIL PROTECTED] wrote: From the Asterisk CLI you can do show queues and show agents. There are also a number of third party tools, free and not-free, to take information from Asterisk and present it in real-time and on a historical basis. AsteriskGuru Queue Statistics http://www.asteriskguru.com/tools/queue_stats.php I'm sure there are others. Maybe someone else can kick in a couple other links/projects? You may want to check our QueueMetrics too - it's a commercial product used in some of the largest Asterisk call centers worldwide but it's free for small CCs, SOHOs and individual hackers. See http://queuemetrics.loway.it Yours l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Recommendations on web interface for IT staff
[EMAIL PROTECTED] wrote: I am proposing an Asterisk system of many servers to service multiple departments in a number of locatations to a large client. They have an IT department but their Linux skills are weak and they are likely to face a high churn rate in staff so it would not be wise to expect a high level of Linux expertise to be maintained. I am thinking it would be best to do the nitty gritty glue work at the config file level myself but have a web based interface to common tasks such as managing extensions, adding trucks, voicemail etc. They are anxious for obvious reasons that they are able to manage the system without having to call me every time they need changes. As it will be a multi-server system there will be some fairly detailed configs to put together, so I would think a [EMAIL PROTECTED] installation would not be suitable, but I haven't tested that theory so I am not against trying it. What recommendations for web management can you make from experience of larger systems? It doesn't have to be limited to free systems. I am also interested in opinions on whether you would implement one monster server to do everything and have parts to maintain it, or would your preference be to have one server per department and interlink them, keeping the hardware the same and having a standby system ready to fill in for failed systems. On one hand there is only one server to monitor, on the other there is redundancy but also complexity. I can see advantages in both approaches. Chris, PBXware comes as standard with the features your client requires: http://www.bicomsystems.com/popup/319/C/features/P_2571/#a1597 If you need more info please contact me! Senad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM2400 wierdness
I also had DTMF problems with my TDM400 when I upgraded from Asterisk 1.0.9 to 1.2.1. After the upgrade I noticed that my bank IVR and work VM would not recognize DTMF coming from my * system. I had to add the 'toneduration' parameter and bump it up to 300ms before it began to work correctly for me. Can someone post what the default 'toneduration' is when not explicitly specified in zapata.conf? - Original Message - From: Roger Hill [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Thursday, December 29, 2005 3:08 AM Subject: Re: TDM2400 wierdness Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.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 -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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] Affordable IP Phones for Asterisk
you want something really cheap. you got to visit www.broad-tel.com. It is even offering a WiFi phone at US$125 for its existing clients. On 12/20/05, Dakota [EMAIL PROTECTED] wrote: Are there any IP Phones that can work with Asterisk, that cost less than $60?if so, what's the model/manufacturer? Dakota___--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] Snom 190 occasionally NR, SIP 401
Snom 190 phone (snom190-SIP 3.60k) occasionally gets SIP 401 response from Asterisk 1.2.1 server. A few minutes later is registered again. It happend at least two times since Asterisk version 1.2.1 is used at the server, but I am not shure if the problem already existed before this update. Has anyone encountered a similar problem? -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Affordable IP Phones for Asterisk
There will be one announced at CES next week by a major company. Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com On 12/20/05, Dakota [EMAIL PROTECTED] wrote: Are there any IP Phones that can work with Asterisk, that cost less than $60?if so, what's the model/manufacturer? Dakota___--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 FXO Panasonic PBX
There are4 options for your consideration: 1. use 2 x 1-port FXO gateway 2. use 2-port FXS gateway with FXS to FXO converter 3. use a 4-port FXO gateway. 4. use 2 x X100P cards You can get them from www.broad-tel.com On 12/21/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I'm looking for a reliable 2 FXO-port gateway to connect a PanasonicPBX to Asterisk. Can anyone recommend a stable and reliable one? Thanks,Waldo___--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] Need HT488 FXO example for both inbound andoutbound.
Hi James, This link might help: https://billing.atlasvoice.com/forum/index.php?topic=20.0 -- Bjorn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Ronald Sent: Sunday, January 01, 2006 1:56 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Need HT488 FXO example for both inbound andoutbound. I'm new to Asterisk and I'm looking for example of how to set up the FXO side of an HT488. I have the FXS side working and can place calls between it and soft phone just fine. What I was able to find the Wiki, forums google has not been useful to me. I think I'm missing something simple probably on the HT488 device. Once I have working example I'd be happy to post it on the Wiki for others. BTW, I purchased the HT488 because I was told it's a direct replacement for the Supra 3000 which is no longer directly available to end users per Cisco. If it's the HT488 that's a piece of junk someone please let me know so I can return it. Thanks James Ronald ___ --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] Re: What is the best Dell Machine for Asterisk?
On Wed, Dec 28, 2005 at 04:02:00PM -0800, William Boehlke wrote: The 830s are nice but limited because they do RAID on a card and have but one suitable PCI slot. So you can have an interface card or RAID, but not both. Linux software raid is, in our experience, much better than any hardware raid solution. We admin 20+ machines all booting on soft raid 1 or 5 partitions up to 2 TB. -- A good friend will help you move, a true friend will help you move a body. ___ --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] Got 200 OK on REGISTER that isn't a register
What does this warning mean? WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on REGISTER that isn't a register ___ --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] Got 200 OK on REGISTER that isn't a register
On 1/1/06, Robert La Ferla [EMAIL PROTECTED] wrote: What does this warning mean? WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on REGISTER that isn't a register Your SIP device is returning a 200 OK message about a registration attempt, but Asterisk doesn't believe there is a registration attempt in progress with this phone. This is what's generating the message. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: GXP-2000 fw 1.0.1.13 and NTP
Leif Neland [EMAIL PROTECTED] writes: My GS BT101 have also developed problems with sync'ing to my ntp-server. I can see, using tcpdump, that the phone asks my server and gets an answer, but the display is not updated. It used to work, but now it usually doesn't, but strangely, sometime it does... Try power cycling the phone. The Grandstreams seem to get flakier the longer they are up. Normally I notice it when they the phones stop allowing incoming www connections. A power cycle always cures it. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --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 HT488 FXO example for both inboundandoutbound.
Bjorn, Thanks!! The example looks like what I need although I don't understand the Forward to VoIP as I don't have an ITSP. I'll give it a try later today. Basic Settings: Number of rings: 1;0 is not a valid option Forward to VoIP: a number in your from-pstn context where you want to receive incoming calls Thanks again and happy new year... JR - Original Message - From: Bjorn Asmul [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 01, 2006 11:55 AM Subject: RE: [Asterisk-Users] Need HT488 FXO example for both inboundandoutbound. Hi James, This link might help: https://billing.atlasvoice.com/forum/index.php?topic=20.0 -- Bjorn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Ronald Sent: Sunday, January 01, 2006 1:56 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Need HT488 FXO example for both inbound andoutbound. I'm new to Asterisk and I'm looking for example of how to set up the FXO side of an HT488. I have the FXS side working and can place calls between it and soft phone just fine. What I was able to find the Wiki, forums google has not been useful to me. I think I'm missing something simple probably on the HT488 device. Once I have working example I'd be happy to post it on the Wiki for others. BTW, I purchased the HT488 because I was told it's a direct replacement for the Supra 3000 which is no longer directly available to end users per Cisco. If it's the HT488 that's a piece of junk someone please let me know so I can return it. Thanks James Ronald ___ --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] Having major issues with TDM2400
Hello Kerry, Maybe it's me, but why are you using hint in this fashion? Shouldn't you be doing exten = 100,1,Dial(SIP/900zap/g0/w5551212) or is there something new that I have missed? Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Saturday, December 31, 2005 11:38 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Having major issues with TDM2400 To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry ___ --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] Having major issues with TDM2400
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, January 01, 2006 11:42 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Having major issues with TDM2400
Hello Kerry,
Maybe it's me, but why are you using hint in this fashion?
Shouldn't you be doing exten =
100,1,Dial(SIP/900zap/g0/w5551212) or is there something new
that I have missed?
Regards,
Greg
I apologize for not being a config-file pureist, but I have this working
just fine on my office machine (using IAX trunks).
Below is the extensions_additional.conf as created by AMP. If there is any
more information I can provide, please ask. -Kerry
[ext-local]
include = ext-local-custom
exten = 100,1,Macro(exten-vm,100,100)
exten = ${VM_PREFIX}100,1,Macro(vm,100)
exten = 100,hint,SIP/900SIP/901zap/g0/w2831212
exten = 101,1,Macro(exten-vm,novm,101)
exten = 101,hint,
exten = 300,1,Macro(exten-vm,novm,300)
exten = 300,hint,SIP/1000SIP/1200zap/g0/w8421212
exten = 301,1,Macro(exten-vm,301,301)
exten = ${VM_PREFIX}301,1,Macro(vm,301)
exten = 301,hint,zap/g0/w9331212SIP/1001SIP/1201
exten = 302,1,Macro(exten-vm,302,302)
exten = ${VM_PREFIX}302,1,Macro(vm,302)
exten = 302,1,Macro(exten-vm,302,302)
exten = ${VM_PREFIX}302,1,Macro(vm,302)
exten = 302,hint,SIP/1002SIP/1202zap/g0/w17149261212
exten = 303,1,Macro(exten-vm,303,303)
exten = ${VM_PREFIX}303,1,Macro(vm,303)
exten = 303,hint,SIP/1003SIP/1203zap/g0/w17143691212
exten = 304,1,Macro(exten-vm,304,304)
exten = ${VM_PREFIX}304,1,Macro(vm,304)
exten = 304,hint,SIP/1004zap/g0/w17144760731
exten = 305,1,Macro(exten-vm,305,305)
exten = ${VM_PREFIX}305,1,Macro(vm,305)
exten = 305,hint,SIP/1005SIP/1205zap/g0/w4331212
exten = 306,1,Macro(exten-vm,306,306)
exten = ${VM_PREFIX}306,1,Macro(vm,306)
exten = 306,hint,SIP/1006SIP/1206zap/g0/6361212
exten = 307,1,Macro(exten-vm,307,307)
exten = ${VM_PREFIX}307,1,Macro(vm,307)
exten = 307,hint,SIP/1007SIP/1207zap/g0/w15627151212
exten = 308,1,Macro(exten-vm,308,308)
exten = 307,hint,SIP/1007SIP/1207zap/g0/w15627151212
exten = 308,1,Macro(exten-vm,308,308)
exten = ${VM_PREFIX}308,1,Macro(vm,308)
exten = 308,hint,zap/g0/w2941212SIP/1008SIP/1208
exten = 309,1,Macro(exten-vm,309,309)
exten = ${VM_PREFIX}309,1,Macro(vm,309)
exten = 309,hint,SIP/1009
exten = 310,1,Macro(exten-vm,310,310)
exten = ${VM_PREFIX}310,1,Macro(vm,310)
exten = 310,hint,SIP/1204SIP/1010
exten = none,hint,
___
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Re: [Asterisk-Users] voicemail/privacy system
Moises Silva wrote: Yep, perfectly possible. I would do that with AGI and php, in your case, perl works as well. The only thing you need is read documentation regarding AGI, Voicemail and extensions. Its kind of difficult to helo you further if you dont tell us how much you know about contexts, extensions etc. But in general you will I've read The ASterisk Handbook Verson 2, so I have a very basic understanding of contexts, extensions and so forth. After reading more on AGI, it looks like I could do everything I want to with a rather simple Asterisk config and a perl script. The privacy manager function could be accomplished by looking for an empty (or missing) agi_callerid: value from STDIN. And the individual voicemail boxes could be accomplished by playing a prompt, waiting for a digit and using set_extension to send the call to the desired voicemail box. Does that sound like I'm on the right track? Thanks in advance, Roy ___ --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] Snom 190 occasionally NR, SIP 401
On 17:29, Sun 01 Jan 06, Stefan Tichy wrote: Snom 190 phone (snom190-SIP 3.60k) occasionally gets SIP 401 response from Asterisk 1.2.1 server. A few minutes later is registered again. It happend at least two times since Asterisk version 1.2.1 is used at the server, but I am not shure if the problem already existed before this update. Has anyone encountered a similar problem? yeah, we have the same trouble using asterisk 1.0.9-bristuff -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Snom 190 occasionally NR, SIP 401
On 21:40, Sun 01 Jan 06, Michiel van Baak wrote: On 17:29, Sun 01 Jan 06, Stefan Tichy wrote: Snom 190 phone (snom190-SIP 3.60k) occasionally gets SIP 401 response from Asterisk 1.2.1 server. A few minutes later is registered again. It happend at least two times since Asterisk version 1.2.1 is used at the server, but I am not shure if the problem already existed before this update. Has anyone encountered a similar problem? yeah, we have the same trouble using asterisk 1.0.9-bristuff Forgot to add this problem occured once we upgraded the phones to 3.6X. the 3.56 firmware didn't have this problem, but that firmware was unable to alter headset volume :( -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Cell phone dock/switch as Asterisk FXO source
Cell Socket is another such product. Current Cell Sockets work with some of Motorola phones. Different systems GSM, CDMA, work somewhat differently regarding callerID and speed dial The original CellSocket worked with certain Nokia phones In the GSM version dialing is similar to the PSTN, but the send function uses the # to start the call. Incoming calls produce a ring signal that should be detected by the FXO card. I use mine as a trunk into my house PBX ( not Asterisk ) but I see no reason why it shouldn't work as well. See the list archives for more comments, and use Google to search on cellsocket John Novack Brian McEntire wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling. I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line? ___ --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 1.2.1 segmentation faulting!...
Yes,
I got the same error when I tried to register my G.729 license.
When you downloaded the patch, are you sure you did that on binary or ascii?
My problem was my download was automatic. I forced to binary and the problem
was fixed. Check the size of the files, on your machine and the ftp site.
Happy new year.
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Peeters (Asterisk)
Sent: Sunday, January 01, 2006 6:28 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk 1.2.1 segmentation faulting!...
I am having issues with 1.2.1/BriStuff 0.3.Pre 1d/Florz patch
On a *very* regular basis I get:
Disconnected from Asterisk server
/usr/sbin/safe_asterisk: line 42: 1359 Segmentation fault
${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Anyone seen this? Any ideas?
TIA BRgds
--
F Peeters
PIII 450 - 1 GB - * 1.2.1 - BRIstuff 0.3.0 Pre 1d - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
___
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Re: [Asterisk-Users] Having major issues with TDM2400
On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a number using the hint priority? The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Which makes sense to me, since as soon as you start dialing you *are* off hook, which in analog means the phone *is* answered. Since all the singalling is done in band, it is not difference than picking up the Zap channel for incoming call, at which point you also understand it's considered answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. So don't use callprogress if it doesn't work for you, in no way do I see this related to the subject line of this post. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Again what is this suppose to do? Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry ___ --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 FXO Panasonic PBX
On 1/1/06, VoIP Newbie [EMAIL PROTECTED] wrote: There are 4 options for your consideration: 1. use 2 x 1-port FXO gateway 2. use 2-port FXS gateway with FXS to FXO converter What is an FXS to FXO converter? you have any URLs? 3. use a 4-port FXO gateway. 4. use 2 x X100P cards You can get them from www.broad-tel.com On 12/21/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic PBX to Asterisk. Can anyone recommend a stable and reliable one? Thanks, Waldo ___ --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] Having major issues with TDM2400
The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a number using the hint priority? The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Which makes sense to me, since as soon as you start dialing you *are* off hook, which in analog means the phone *is* answered. Since all the singalling is done in band, it is not difference than picking up the Zap channel for incoming call, at which point you also understand it's considered answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. So don't use callprogress if it doesn't work for you, in no way do I see this related to the subject line of this post. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Again what is this suppose to do? Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk FXO Panasonic PBX
Waldo Rubinstein wrote: I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic PBX to Asterisk. Can anyone recommend a stable and reliable one? Use 2x Sipura SPA-3000 - and you will also get 2x FXS... Or use a Digium TDM02B (2x FXO). ___ --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] Having major issues with TDM2400
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry Kerry, You don't get call progress with FXO zap channels as you would with VoIP or PRI. That being the case, the FXO port never signals that it is ringing with call progress, but rather, goes to an up state (answered) as soon as it's finished dialing, whether the remote end has answered or not. You're going to be hard pressed to have a ZAP channel and SIP channel trying to detect who's going to answer first, because the Zap/FXO port is always going to win and it's not going to be because it's always answered. -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [Asterisk-Users] Having major issues with TDM2400
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 How? using Zap FXS? or Zap FXO? The question has been answered by me and BJ, You will not get status of the POTS using Zap, because it's already answered as soon as you take it off hook, some good workaround examples exist in the user list archive, amongst them: * Implement a macro using the M option in the dial command to not bridge the call until a certain key is pressed. * Implement the c option for the zap channel. Again this is NOT a problem with Digium/TDM2400/Asterisk/Zaptel, but with you reposting the same question after it has been answered, maybe you should not use AMP but Asterisk from source then you will understand this better. I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a number using the hint priority? The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Which makes sense to me, since as soon as you start dialing you *are* off hook, which in analog means the phone *is* answered. Since all the singalling is done in band, it is not difference than picking up the Zap channel for incoming call, at which point you also understand it's considered answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. So don't use callprogress if it doesn't work for you, in no way do I see this related to the subject line of this post. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Again what is this suppose to do? Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Having major issues with TDM2400
Hello Kerry, I do it exactly as such, however in steps. My understanding of the hint system is just for notification of status, not for execution of dialing. I regularly use this same setup you are looking for, rings in, then rings 2-5 devices (some zap, some iax) and the first one that answers gets the call. Make sure you use the Dial( command I replied with previously. (avoid hint for testing). Looking at your emails, it looks like you need to review the dialplan setup, for example the hint and do not look right to me. One example for me: exten = s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,) But it is the same as SIP/220Zap/5, etc. I cannot say anything specific to amp however. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 7:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a number using the hint priority? The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Which makes sense to me, since as soon as you start dialing you *are* off hook, which in analog means the phone *is* answered. Since all the singalling is done in band, it is not difference than picking up the Zap channel for incoming call, at which point you also understand it's considered answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. So don't use callprogress if it doesn't work for you, in no way do I see this related to the subject line of this post. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Again what is this suppose to do? Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Having major issues with TDM2400
Oh just a followup, if you are trying to do an outbound dialout over analog, what others are saying is correct. You could consider however using a voip provider to make the outbound call, then you should have status. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Sunday, January 01, 2006 8:05 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Hello Kerry, I do it exactly as such, however in steps. My understanding of the hint system is just for notification of status, not for execution of dialing. I regularly use this same setup you are looking for, rings in, then rings 2-5 devices (some zap, some iax) and the first one that answers gets the call. Make sure you use the Dial( command I replied with previously. (avoid hint for testing). Looking at your emails, it looks like you need to review the dialplan setup, for example the hint and do not look right to me. One example for me: exten = s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,) But it is the same as SIP/220Zap/5, etc. I cannot say anything specific to amp however. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 7:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a number using the hint priority? The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Which makes sense to me, since as soon as you start dialing you *are* off hook, which in analog means the phone *is* answered. Since all the singalling is done in band, it is not difference than picking up the Zap channel for incoming call, at which point you also understand it's considered answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. So don't use callprogress if it doesn't work for you, in no way do I see this related to the subject line of this post. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Again what is this suppose to do? Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --
RE: [Asterisk-Users] Having major issues with TDM2400
Thanks everyone, the reason I posted here was because a Digium support tech said it should work and he couldn't figure it out. So while I appreciate everyone's comments that it wont work, a technician from Digium said it should, hence I turned to the list for clarification. This is not really a good answer for me to go back to my client with as this is one primary feature he liked which pushed him into an Asterisk solution. For right now, their bandwidth is insuffecient for using a SIP provider, although a T1 line is on order. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 01, 2006 5:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Oh just a followup, if you are trying to do an outbound dialout over analog, what others are saying is correct. You could consider however using a voip provider to make the outbound call, then you should have status. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Sunday, January 01, 2006 8:05 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Hello Kerry, I do it exactly as such, however in steps. My understanding of the hint system is just for notification of status, not for execution of dialing. I regularly use this same setup you are looking for, rings in, then rings 2-5 devices (some zap, some iax) and the first one that answers gets the call. Make sure you use the Dial( command I replied with previously. (avoid hint for testing). Looking at your emails, it looks like you need to review the dialplan setup, for example the hint and do not look right to me. One example for me: exten = s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,) But it is the same as SIP/220Zap/5, etc. I cannot say anything specific to amp however. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 7:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a number using the hint priority? The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Which makes sense to me, since as soon as you start dialing you *are* off hook, which in analog means the phone *is* answered. Since all the singalling is done in band, it is not difference than picking up the Zap channel for incoming call, at which point you also understand it's considered answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. So don't use callprogress if it doesn't work for you, in no way do I see this related to the subject line of this post. And using this: exten = 100,hint,SIP/900zap/g0/w5551212 Again what is this suppose to do? Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges
Re: [Asterisk-Users] Having major issues with TDM2400
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks everyone, the reason I posted here was because a Digium support tech said it should work and he couldn't figure it out. So while I appreciate everyone's comments that it wont work, a technician from Digium said it should, hence I turned to the list for clarification. This is not really a good answer for me to go back to my client with as this is one primary feature he liked which pushed him into an Asterisk solution. For right now, their bandwidth is insuffecient for using a SIP provider, although a T1 line is on order. The syntax does work when that Zap channel is part of a PRI span, it just will not work with FXO. If you have an issue with the support provided by Digium staff, I strongly recommend you take that up with them directly. I've always found them to be very receptive to making sure the right information gets out if there was a mistake made on their part. -- Bird's The Word Technologies, Inc. http://www.btwtech.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
RE: [Asterisk-Users] Having major issues with TDM2400
I did not say I had a problem with support. The problem was the tech ran out of time on Friday and there was nobody to escalate the problem to. So instead of waiting until tomorrow for teir 2 support, I looked to the people on the list to see if I could find an answer before then. It seems as though I have got an answer and I will verify with them tomorrow. Unfortunatly it is bad news for the client. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Sunday, January 01, 2006 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks everyone, the reason I posted here was because a Digium support tech said it should work and he couldn't figure it out. So while I appreciate everyone's comments that it wont work, a technician from Digium said it should, hence I turned to the list for clarification. This is not really a good answer for me to go back to my client with as this is one primary feature he liked which pushed him into an Asterisk solution. For right now, their bandwidth is insuffecient for using a SIP provider, although a T1 line is on order. The syntax does work when that Zap channel is part of a PRI span, it just will not work with FXO. If you have an issue with the support provided by Digium staff, I strongly recommend you take that up with them directly. I've always found them to be very receptive to making sure the right information gets out if there was a mistake made on their part. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Having major issues with TDM2400
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks everyone, the reason I posted here was because a Digium support tech said it should work and he couldn't figure it out. So while I appreciate everyone's comments that it wont work, a technician from Digium said it should, hence I turned to the list for clarification. This is not really a good answer for me to go back to my client with as this is one primary feature he liked which pushed him into an Asterisk solution. For right now, It will still work using the M option in the dial command, as I wrote before, also look up the follwoing: http://www.voip-info.org/wiki-asterisk+cmd+dial http://bugs.digium.com/view.php?id=5574 Using some creativity you can give your client what you promised plus. their bandwidth is insuffecient for using a SIP provider, although a T1 line is on order. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 01, 2006 5:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Oh just a followup, if you are trying to do an outbound dialout over analog, what others are saying is correct. You could consider however using a voip provider to make the outbound call, then you should have status. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Sunday, January 01, 2006 8:05 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Hello Kerry, I do it exactly as such, however in steps. My understanding of the hint system is just for notification of status, not for execution of dialing. I regularly use this same setup you are looking for, rings in, then rings 2-5 devices (some zap, some iax) and the first one that answers gets the call. Make sure you use the Dial( command I replied with previously. (avoid hint for testing). Looking at your emails, it looks like you need to review the dialplan setup, for example the hint and do not look right to me. One example for me: exten = s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,) But it is the same as SIP/220Zap/5, etc. I cannot say anything specific to amp however. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 7:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a number using the hint priority? The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Which makes sense to me, since as soon as you start dialing you *are* off hook, which in analog means the phone *is* answered. Since all the singalling is done in band, it is not difference than picking up the Zap channel for incoming call, at which point you also understand it's considered answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing
Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
Are you using raid for performance or redundancy? Software raid is nice except when the drive that fails is the one with your boot partition on it. I guess you could always tftp boot the kernel or something. Craig - Original Message - From: Louis-David Mitterrand [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 02, 2006 1:17 AM Subject: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk? On Wed, Dec 28, 2005 at 04:02:00PM -0800, William Boehlke wrote: The 830s are nice but limited because they do RAID on a card and have but one suitable PCI slot. So you can have an interface card or RAID, but not both. Linux software raid is, in our experience, much better than any hardware raid solution. We admin 20+ machines all booting on soft raid 1 or 5 partitions up to 2 TB. -- A good friend will help you move, a true friend will help you move a body. ___ --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] Having major issues with TDM2400
As much as I like the option of implementing a follow-me type of script, the second problem is that the client wants to use AMP to manage the extensions. Just doesn't seem like I have a solution that fits all of the client's requirements. The easiest solution seems to be to use a SIP trunk for the outbound call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 6:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks everyone, the reason I posted here was because a Digium support tech said it should work and he couldn't figure it out. So while I appreciate everyone's comments that it wont work, a technician from Digium said it should, hence I turned to the list for clarification. This is not really a good answer for me to go back to my client with as this is one primary feature he liked which pushed him into an Asterisk solution. For right now, It will still work using the M option in the dial command, as I wrote before, also look up the follwoing: http://www.voip-info.org/wiki-asterisk+cmd+dial http://bugs.digium.com/view.php?id=5574 Using some creativity you can give your client what you promised plus. their bandwidth is insuffecient for using a SIP provider, although a T1 line is on order. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 01, 2006 5:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Oh just a followup, if you are trying to do an outbound dialout over analog, what others are saying is correct. You could consider however using a voip provider to make the outbound call, then you should have status. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Sunday, January 01, 2006 8:05 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Hello Kerry, I do it exactly as such, however in steps. My understanding of the hint system is just for notification of status, not for execution of dialing. I regularly use this same setup you are looking for, rings in, then rings 2-5 devices (some zap, some iax) and the first one that answers gets the call. Make sure you use the Dial( command I replied with previously. (avoid hint for testing). Looking at your emails, it looks like you need to review the dialplan setup, for example the hint and do not look right to me. One example for me: exten = s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,) But it is the same as SIP/220Zap/5, etc. I cannot say anything specific to amp however. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 7:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten = 100,hint,SIP/900zap/g0/w5551212 What are you trying to do here? You trying to hint to a zip channel and dial a
RE: [Asterisk-Users] Having major issues with TDM2400
Perhaps a Sipura-3000 could be of use here? Anyone have any ideas about that? Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 10:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 As much as I like the option of implementing a follow-me type of script, the second problem is that the client wants to use AMP to manage the extensions. Just doesn't seem like I have a solution that fits all of the client's requirements. The easiest solution seems to be to use a SIP trunk for the outbound call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 6:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks everyone, the reason I posted here was because a Digium support tech said it should work and he couldn't figure it out. So while I appreciate everyone's comments that it wont work, a technician from Digium said it should, hence I turned to the list for clarification. This is not really a good answer for me to go back to my client with as this is one primary feature he liked which pushed him into an Asterisk solution. For right now, It will still work using the M option in the dial command, as I wrote before, also look up the follwoing: http://www.voip-info.org/wiki-asterisk+cmd+dial http://bugs.digium.com/view.php?id=5574 Using some creativity you can give your client what you promised plus. their bandwidth is insuffecient for using a SIP provider, although a T1 line is on order. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 01, 2006 5:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Oh just a followup, if you are trying to do an outbound dialout over analog, what others are saying is correct. You could consider however using a voip provider to make the outbound call, then you should have status. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Sunday, January 01, 2006 8:05 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Hello Kerry, I do it exactly as such, however in steps. My understanding of the hint system is just for notification of status, not for execution of dialing. I regularly use this same setup you are looking for, rings in, then rings 2-5 devices (some zap, some iax) and the first one that answers gets the call. Make sure you use the Dial( command I replied with previously. (avoid hint for testing). Looking at your emails, it looks like you need to review the dialplan setup, for example the hint and do not look right to me. One example for me: exten = s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,) But it is the same as SIP/220Zap/5, etc. I cannot say anything specific to amp however. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 7:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote: To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So
RE: [Asterisk-Users] Having major issues with TDM2400
Well, it would have to be 4 of them for each of the available PSTN lines. I have also considered a Mediatrix channel bank. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 01, 2006 7:53 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Perhaps a Sipura-3000 could be of use here? Anyone have any ideas about that? Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 10:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 As much as I like the option of implementing a follow-me type of script, the second problem is that the client wants to use AMP to manage the extensions. Just doesn't seem like I have a solution that fits all of the client's requirements. The easiest solution seems to be to use a SIP trunk for the outbound call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, January 01, 2006 6:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Having major issues with TDM2400 On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks everyone, the reason I posted here was because a Digium support tech said it should work and he couldn't figure it out. So while I appreciate everyone's comments that it wont work, a technician from Digium said it should, hence I turned to the list for clarification. This is not really a good answer for me to go back to my client with as this is one primary feature he liked which pushed him into an Asterisk solution. For right now, It will still work using the M option in the dial command, as I wrote before, also look up the follwoing: http://www.voip-info.org/wiki-asterisk+cmd+dial http://bugs.digium.com/view.php?id=5574 Using some creativity you can give your client what you promised plus. their bandwidth is insuffecient for using a SIP provider, although a T1 line is on order. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 01, 2006 5:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Oh just a followup, if you are trying to do an outbound dialout over analog, what others are saying is correct. You could consider however using a voip provider to make the outbound call, then you should have status. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Sunday, January 01, 2006 8:05 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Having major issues with TDM2400 Hello Kerry, I do it exactly as such, however in steps. My understanding of the hint system is just for notification of status, not for execution of dialing. I regularly use this same setup you are looking for, rings in, then rings 2-5 devices (some zap, some iax) and the first one that answers gets the call. Make sure you use the Dial( command I replied with previously. (avoid hint for testing). Looking at your emails, it looks like you need to review the dialplan setup, for example the hint and do not look right to me. One example for me: exten = s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,) But it is the same as SIP/220Zap/5, etc. I cannot say anything specific to amp however. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, January 01, 2006 7:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Having major issues with TDM2400 The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry -Original Message- From:
Re: [Asterisk-Users] name that vendor...
[EMAIL PROTECTED] wrote: Well yeah, I had no intention of buying one, I was just wondering what the hell it actually was that the seller was trying to hide. Their supplier? ___ --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] (Fwd) hi there
www.antek.com.tw Had 4 port fxo, for around 200 to 250$ They are OEM, and can change things if u need. I tested it breifly in there office last year in Computex 2005 You can contact [EMAIL PROTECTED] for wholesale. Rehan On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? I bet it has an fcc id which can be looked up at fcc.gov. If you get the first 3 letters it tells you who the vendor is. Maybe a ruse about not believing that it has all those compliance certifications and you want to guarantee the FCC certification for use in the US ... I would google for the name on the sticker, which is 'fxo-04'. This returns people talking about teh Asotel(Dinamyx) fxo-04. There is also a 'stargate fxo-04'. On and on ... If I had to guess I would say it looks like: http://www.chinanetphone.com/newchanpin/fxo-04.asp or http://www.repotec.com/voip/RP_FXO02A.htm My guess is that you should be able to find out more on your own :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group --- End of forwarded message --- --- End of forwarded message --- Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --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] What is the best Dell Machine for Asterisk?
Hiu Yen Onn wrote: How big of RAM for Asterisk server? My production environment will be about 400 users in the office. In one server? 4GB. And more if you can. I'd suggest you use several servers for 400 users unless the percentage of active phones is ~10%. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommendations on web interface for IT staff
Chris Mason (Lists) wrote: I am proposing an Asterisk system of many servers to service multiple departments in a number of locatations to a large client. They have an IT department but their Linux skills are weak and they are likely to face a high churn rate in staff so it would not be wise to expect a high level of Linux expertise to be maintained. http://freshmeat.net/search/?q=asterisksection=projectsGo.x=0Go.y=0 These projects from the URL above should be helpful: http://freshmeat.net/projects/amportal/ http://freshmeat.net/projects/acami/ http://freshmeat.net/projects/astbill/ http://freshmeat.net/projects/astguiclient/ Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there
Hi, Not very reliable for commercial setups, they do have issues hanging up ports etc. Quintum over Antek any day. Regards, Sahil Gupta VoiceValley On Mon, 2 Jan 2006, Rehan AllahWala wrote: www.antek.com.tw Had 4 port fxo, for around 200 to 250$ They are OEM, and can change things if u need. I tested it breifly in there office last year in Computex 2005 You can contact [EMAIL PROTECTED] for wholesale. Rehan On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? I bet it has an fcc id which can be looked up at fcc.gov. If you get the first 3 letters it tells you who the vendor is. Maybe a ruse about not believing that it has all those compliance certifications and you want to guarantee the FCC certification for use in the US ... I would google for the name on the sticker, which is 'fxo-04'. This returns people talking about teh Asotel(Dinamyx) fxo-04. There is also a 'stargate fxo-04'. On and on ... If I had to guess I would say it looks like: http://www.chinanetphone.com/newchanpin/fxo-04.asp or http://www.repotec.com/voip/RP_FXO02A.htm My guess is that you should be able to find out more on your own :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group --- End of forwarded message --- --- End of forwarded message --- Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there
Does Quintum has a 4 port fxo box ? Hi, Not very reliable for commercial setups, they do have issues hanging up ports etc. Quintum over Antek any day. Regards, Sahil Gupta VoiceValley On Mon, 2 Jan 2006, Rehan AllahWala wrote: www.antek.com.tw Had 4 port fxo, for around 200 to 250$ They are OEM, and can change things if u need. I tested it breifly in there office last year in Computex 2005 You can contact [EMAIL PROTECTED] for wholesale. Rehan On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? I bet it has an fcc id which can be looked up at fcc.gov. If you get the first 3 letters it tells you who the vendor is. Maybe a ruse about not believing that it has all those compliance certifications and you want to guarantee the FCC certification for use in the US ... I would google for the name on the sticker, which is 'fxo-04'. This returns people talking about teh Asotel(Dinamyx) fxo-04. There is also a 'stargate fxo-04'. On and on ... If I had to guess I would say it looks like: http://www.chinanetphone.com/newchanpin/fxo-04.asp or http://www.repotec.com/voip/RP_FXO02A.htm My guess is that you should be able to find out more on your own :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group --- End of forwarded message --- --- End of forwarded message --- Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --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] Codec
Hi I am trying to use g.726 so as to make calls, further i am using cisco PAP ATA's, on these PAP's i have a number of options ranging from 16 to 64 kbps for g.726, i wud prefer to use the 16 kbps version, as in it is my sip.conf i have done this allow=g726 On the PAPS i have selected g.726-32 version (at present), now when i am making my calls my call is straight going into voicemail, I wud prefer to use as mentioned above the 16 kbps version but for now i wud give anything to just make it( the entire asterisk Linksys PAP2NA package) work with g.726. I cannot find any examples for g.726 in sip.conf so i dont really have an idea!!! Regards Desperate hrishi ___ --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 Christmas Help request
5) How do I change the time zone for Asterisk? Currently the system time is correct but when I dial *60 it reports a different time (out by many hours). I'm not familiar with this option. Can you please tell me more or send me some link. FYI, this is the relevant extensions_custom.conf entry on an AAH system: exten = *60,1,Answer exten = *60,2,Playback(at-tone-time-exactly) exten = *60,3,SayUnixTime(,,IMp) exten = *60,4,Playback(beep) exten = *60,5,Hangup [Description] SayUnixTime([unixtime][|[timezone][|format]]) unixtime: time, in seconds since Jan 1, 1970. May be negative. defaults to now. timezone: timezone, see /usr/share/zoneinfo for a list. defaults to machine default. format: a format the time is to be said in. See voicemail.conf. defaults to "ABdY 'digits/at' IMp" -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
