Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices
The legal and medical communities still seem to prefer faxing, in the ( mistaken? ) belief that it is more secure. In fact the medical community is fearful of the legal beagles. These groups are really slow to change. At least in the USA The couple of times I have received medical faxes to my fax bank scare me about the actual security. My company is not in the medical field, nowhere close, in fact. In one case, the fax included the patients name, address, phone, Date of Birth, SSN, and confidential medical history. The comment I made to a coworker was that if I wanted to steal an identity, they had just handed me everything I would need. In the second case, it was a question from a pharmacy to a doctors office. Not quite so bad. I called up the pharmacy and said I had a problem with a fax they had sent. After asking me for some information from the fax so they could identify which patient I was calling about they asked what the problem was. I replied that I was a manufacturer of balloons and not a doctor's office. To say there was a bit of panic creeping into the guys voice on the other end was an understatement. I think I triggered some HIPAA reporting provisions.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices
On 15-06-15 08:48 PM, Matt Darnell wrote: In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built into the unit. Unfortunately the FXS ports are just ATA's off of Asterisk, no different than a SPA2012 unit. The 550 is getting long in the tooth and very expensive for a few FXS ports, what are you folks doing when someone has a need? It can be a modem for the power company to read the meter, a postage machine that needs to get more postage, an alarm system,etc. Finding less and less need for analog circuits. Alarm - envisalink, postage machine - ethernet based, pos terminal - ethernet. Fax is really the only need recently, and even that has alternatives like emailing scans that most people prefer now. Is the customer buying a POTS line and splitting it the only other way? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices
Jon Pounder wrote: snip Fax is really the only need recently, and even that has alternatives like emailing scans that most people prefer now. The legal and medical communities still seem to prefer faxing, in the ( mistaken? ) belief that it is more secure. In fact the medical community is fearful of the legal beagles. These groups are really slow to change. At least in the USA John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices
The mediatrix 4102s line kicks ass. On Jun 15, 2015 8:49 PM, Matt Darnell mattdarn...@gmail.com wrote: In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built into the unit. Unfortunately the FXS ports are just ATA's off of Asterisk, no different than a SPA2012 unit. The 550 is getting long in the tooth and very expensive for a few FXS ports, what are you folks doing when someone has a need? It can be a modem for the power company to read the meter, a postage machine that needs to get more postage, an alarm system,etc. Is the customer buying a POTS line and splitting it the only other way? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS hangup issues
I currently have an Asterisk 1.8.8.1 system set up with SIP accounts as well as a Wildcard TDM400P REV I card with both FXS and FXO ports - FXO is connected to outside lines, FXS connected to inside analog phones. Everything about the setup works fine except one thing - after making calls to or from any of the analog phones, and the other side hangs up, the analog phone just gives a busy signal instead of hanging up. On the Asterisk console, it seems to think it's hung up the phone too: == Spawn extension (from-office, 44, 50005) exited non-zero on 'DAHDI/5-1' -- Hanging up on 'DAHDI/5-1' -- Hungup 'DAHDI/5-1' chan_dahdi.conf is mostly just the default with just the lines defined, nothing too fancy, and this doesn't happen for SIP clients or remote phones via the FXO ports. Any ideas? You have to hang up the phone too. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS ports on TDM410P card...
[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006- of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid extension 's' in context 'default', but no invalid handler
--
I have the code to set up an extension for toggling Telco pass through
working I think. What isn't working is the pass through. I get the
above error messages when I try to call the POTS line connected to
DAHDI/1 from my Comcast line.
I'm noticing other warning messages cropping up about this file or that
file not existing and modules not loading, but mostly the system seems
to be working so I'm wondering if these warnings are relevant. I'm
using Asterisk 1.8.
I think that [from-pstn] isn't working...
For those who don't know what I'm after, I'm trying when a phone company
call comes in to ring SIP phones and local FXS lines on my TDM410P. The
purpose of the toggle is to be able to disable this feature. Sometimes,
I really want to use this system as a private intercom system where at
other times, ringing remote SIP phones for an incoming telephone company
call might be needed. Say you are at extension 2000 or 2002, SIP phones
in other buildings, and you want or need to be able to receive calls
from the PSTN.
I'm in the U.S., under the [external] section am I blocking long
distance outgoing phone calls? In the U.S., you dial 1 and then
the number for long distance. Essentially, what I need to do is
block dialing 1 and then a number with the exception of 1-800 or
1-866.
Thank you for taking the time to look at my questions and
information ;-)
My current extensions.conf file in it's entirety follows:
-
[globals]
CENTURYLINK=DAHDI/1
COMCAST=DAHDI/2
ANDREWROOM=DAHDI/3
SERVERROOM=DAHDI/4
WIDE_PBX=SIP/2000SIP/2002SIP/2006SIP/2007SIP/2008SIP/2009
${SERVERROOM}${ANDREWROOM}
INSIDE_PBX=SIP/2006SIP/2007SIP/2008SIP/2009${SERVERROOM}
${ANDREWROOM}
OUTSIDE_PBX=SIP/2000SIP/2002
TELCO_ON=0
PSTN_THROUGH=${SERVERROOM}${ANDREWROOM}SIP/2000SIP/2002
[external]
exten = _9NXXNXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _8NXXNXX,1,Dial(${COMCAST}/${EXTEN:1})
[my-phones]
exten = i,1,Playback(/var/lib/asterisk/sounds/custom/extns-list)
exten = i,n,Hangup()
exten = 2000,1,Dial(SIP/2000,40)
same = n,VoiceMail(2000,u)
exten = 2002,1,Dial(SIP/2002,40)
same = n,VoiceMail(2002,u)
exten = 2004,1,Dial(SIP/2004,40)
same = n,VoiceMail(2004,u)
exten = 2006,1,Dial(SIP/2006,40)
same = n,VoiceMail(2006,u)
exten = 2007,1,Dial(SIP/2007,40)
same = n,VoiceMail(2007,u)
exten = 2008,1,Dial(SIP/2008,40)
same = n,VoiceMail(2008,u)
exten = 2009,1,Dial(SIP/2009,40)
same = n,VoiceMail(2009,u)
exten = 2010,1,Dial(${SERVERROOM},40)
same = n,VoiceMail(2009,u)
exten = 2010,1,Dial(${SERVERROOM},40)
same = n,VoiceMail(2010,u)
exten = 2011,1,Dial(${ANDREWROOM},40)
same = n,VoiceMail(2011,u)
exten = 2012,1,Dial(${WIDE_PBX},40)
exten = 2013,1,Dial(${INSIDE_PBX},40)
exten = 2014,1,Dial(${OUTSIDE_PBX},40)
exten = 2015,1,GoToIf($[${TELCO_ON}=1]?2:5)
; 2 turns off telco_on
exten = 2015,2,Set(GLOBAL(TELCO_ON)=0)
exten = 2015,3,Playback(/var/lib/asterisk/sounds/custom/telco-off)
exten = 2015,4,hangup()
; 5 turns on telco_on
exten = 2015,5,Set(GLOBAL(TELCO_ON)=1)
exten = 2015,6,Playback(/var/lib/asterisk/sounds/custom/telco-on)
exten = 2015,7,hangup()
exten = 2999,1,VoiceMailMain(${CALLERID(num)},s)
[from-pstn]
exten = s,1,GoToIf($[${TELCO_ON}=1]?2:3)
; 2 rings all phones
exten = s,2,Dial(${PSTN_THROUGH},40)
exten = s,3,Hangup()
include = external
include = from-pstn
--
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Re: [asterisk-users] FXS ports on TDM410P card...
On 10/15/2011 05:31 AM, Michael C. Robinson wrote:
[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006- of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid extension 's' in context 'default', but no invalid handler
--
I have the code to set up an extension for toggling Telco pass through
working I think. What isn't working is the pass through. I get the
above error messages when I try to call the POTS line connected to
DAHDI/1 from my Comcast line.
I'm noticing other warning messages cropping up about this file or that
file not existing and modules not loading, but mostly the system seems
to be working so I'm wondering if these warnings are relevant. I'm
using Asterisk 1.8.
I think that [from-pstn] isn't working...
You're not landing in [from-pstn]. Incoming calls are landing in
[default]. That's not a problem in extensions.conf, that's a problem in
dahdi.conf for those channels. They're not in the right context.
For those who don't know what I'm after, I'm trying when a phone company
call comes in to ring SIP phones and local FXS lines on my TDM410P. The
purpose of the toggle is to be able to disable this feature. Sometimes,
I really want to use this system as a private intercom system where at
other times, ringing remote SIP phones for an incoming telephone company
call might be needed. Say you are at extension 2000 or 2002, SIP phones
in other buildings, and you want or need to be able to receive calls
from the PSTN.
I'm in the U.S., under the [external] section am I blocking long
distance outgoing phone calls? In the U.S., you dial 1 and then
the number for long distance. Essentially, what I need to do is
block dialing 1 and then a number with the exception of 1-800 or
1-866.
[external]
exten = _91800NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91888NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91877NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91866NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91NXXNXX,1,Playback(im-sorry-Dave-you-cant-call-that)
exten = _91NXXNXX,n,Congestion
exten = _81800NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81888NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81877NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81866NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81NXXNXX,1,Playback(im-sorry-Dave-you-cant-call-that)
exten = _81NXXNXX,n,Congestion
That'll let you dial US toll free numbers out the channel specified by
dialing 9 or 8. It will playback a message and then generate a
congestion tone if some other number is dialed.
--
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Re: [asterisk-users] FXS ports on TDM410P card...
On Wednesday 12 October 2011, Michael C. Robinson wrote:
[stuff deleted]
Seems to be working now, good eyes.
I have the O'Reilly book on Asterisk 1.8, though I'm wondering where to
look in it to learn how to allow incoming calls from the phone company
to ring SIP box and FXS connected handsets? This would be a neat
feature, especially if there was a way from the handset to turn it off.
You already set a default context for incoming calls from the phone company.
You just need to specify multiple extensions in your Dial() statement,
delimited by and signs. For instance:
exten = s,1,Dial(DAHDI/3DAHDI/4SIP/301,60)
Or you can even create a global variable with the group of phones you want to
ring. For instance
[globals]
ANALOGUE=DAHDI/3DAHDI/4
BATPHONE=SIP/301
and later you can use something like
Dial(${ANALOGUE}${BATPHONE},60)
Just to make sure I'm clear, imagine you are hooked to a Linksys PAP2
and not only want to call out via the phone company but you want to
receive calls there from the phone company as well. Imagine there is
an extension to call that toggles this behavior on/off. So say 2025 is
the special extension which you call and a voice says relaying phone
company on. You hang up then, the phone rings, and you pick up a call
from somewhere remote via the phone company. You hang up when you're
done and decide the behavior should be turned off, so you dial 2025
again and the voice says relaying phone company off. Now if a call is
incoming from the phone company, your phone doesn't ring. You can call
all local extensions and even remote numbers, but you can't receive
remote calls.
You have to set a global variable when your special extension is dialled.
Then you can use GoToIf() to make decisions based on the value.
[globals]
TELCO_ON=1
ALL_PHONES=DAHDI/3DAHDI/4SIP/301
; (assuming 301 is the SIP extension you want to ring)
; .
[internal]
; .
exten = 2025,1,GoToIf($[${TELCO_ON}=1]?2:5)
; 2 turns off telco_on
exten = 2025,2,Set(GLOBAL(TELCO_ON)=0)
exten = 2025,3,Playback(telco-off)
exten = 2025,4,hangup()
; 5 turns on telco_on
exten = 2025,5,Set(GLOBAL(TELCO_ON)=1)
exten = 2025,6,Playback(telco-on)
exten = 2025,7,hangup()
; .
[from-pstn]
exten = s,1,GoToIf($[${TELCO_ON}=1]?2:3)
; 2 rings all phones
exten = s,2,Dial(${ALL_PHONES},60)
; 3 goes to voicemail (assuming you've configured VM on ext 301 .)
exten = s,3,VoiceMail(301,u)
exten = s,4,Hangup()
[stuff deleted] Another neat trick would be to list
what the extensions are when someone enters an invalid extension. Say
someone dials 1011, not one of my extensions and not a remote phone
number prefixed by 8 or 9.
Then you need to look at the i extension, which is called when someone dials
an invalid extension number. Record yourself a suitable message (cheating
way is to leave a voicemail message, which will already be in the format you
want, and cp the file across) and put something in your context like
exten = i,1,Playback(extns-list)
exten = i,n,Hangup()
The last trick I want to pull, I want an extension that will ring
inclusively 2000 to 2011, say 2012. How do I set this up by hand?
Again, use the sign notation for ringing multiple phones:
exten =
2012,1,Dial(SIP/2000SIP/2001SIP/2002SIP/2003SIP/2004SIP/2005SIP/2006SIP/2007SIP/2008SIP/2009SIP/2010SIP/2011,60)
exten = 2012,2,Hangup()
My personal preference is to split departments on the hundreds, and use x00 as
a ring all phones in department number. For instance if numbers like 2xx
are sales, 3xx are purchasing, 4xx are accounts, 5xx are IT, 6xx are factory
floor, then I would make the number to call everybody in accounts 400.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] FXS ports on TDM410P card...
Quoting Michael C. Robinson plu...@robinson-west.com: My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? You appear to have an inconsistency with context names... [root@robin asterisk]# cat chan_dahdi.conf ... [phone](!) ... context = myphones extensions.conf: ... [my-phones] Put that right and it should work, as you've designed it so far. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com This message was sent using IMP, the Internet Messaging Program. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS ports on TDM410P card...
On Wednesday 12 October 2011, Michael C. Robinson wrote: My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? If you can't make calls *from* a phone, but you can make calls *to* it, that suggests a problem with its default context. Your configuration snippets shew myphones as the default context in chan_dahdi.conf, but the context in the dialplan was my-phones. Make them match up, reload all configuration files and it should all Just Work. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS ports on TDM410P card...
Changes so far: chan_dahdi.conf: [my-phones](!) . . . context = my-phones signalling = fxo_ks . . . [phone1](my-phones) . . . [phone2](my-phones) . . . [phone3](my-phones) . . . [phone4](my-phones) And extensions.conf is the same. Seems to be working now, good eyes. I have the O'Reilly book on Asterisk 1.8, though I'm wondering where to look in it to learn how to allow incoming calls from the phone company to ring SIP box and FXS connected handsets? This would be a neat feature, especially if there was a way from the handset to turn it off. Just to make sure I'm clear, imagine you are hooked to a Linksys PAP2 and not only want to call out via the phone company but you want to receive calls there from the phone company as well. Imagine there is an extension to call that toggles this behavior on/off. So say 2025 is the special extension which you call and a voice says relaying phone company on. You hang up then, the phone rings, and you pick up a call from somewhere remote via the phone company. You hang up when you're done and decide the behavior should be turned off, so you dial 2025 again and the voice says relaying phone company off. Now if a call is incoming from the phone company, your phone doesn't ring. You can call all local extensions and even remote numbers, but you can't receive remote calls. Another trick I want to pull is this. I have a few extensions, 2000 to 2011, where I'd like to have an extension someone can call to figure out what these extensions are. Say 1000 or even 0 if that will work. Something easy to remember anyways. Another neat trick would be to list what the extensions are when someone enters an invalid extension. Say someone dials 1011, not one of my extensions and not a remote phone number prefixed by 8 or 9. The last trick I want to pull, I want an extension that will ring inclusively 2000 to 2011, say 2012. How do I set this up by hand? Thank you again for helping me figure out the context problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS to SIP gateway
jonas kellens wrote: Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? I hope to see more replies because I was in your situation some years ago. I'm far from an expert, but in my experience, at _that_ price range you don't have a lot of products to choose from, the Cisco SPA3102 is similar to what you are describing (Plus it's also a PSTN GW). Of this kind, I used GXW-4004 and Linksys PAP2, SPA3000 (the Cisco 3102 predecesor), also some larger AddPac ATAs (www.addpac.com) with excellent results, all of them have their pros and cons. The Digium TDM410 cards and they have a very good price/quality relation, plus they are intended for asterisk, plus you are supporting asterisk development. For an ATA (FXS only) list you can check http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS to SIP gateway
On Wed, Oct 14, 2009 at 12:27 AM, jonas kellens jonas.kell...@telenet.be wrote: Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? We have used Cisco 2800 series routers with voice cards that work fine for PRI, but don't implement SIP the way they should. Analog was not so great. We also tried to convert some Cisco VG-224s to SIP with limited success. I don't recommend using either of those (plus they are expensive...) Grandstream (GXW-4024) had major issues with Fax, so we only use them for connecting for voice only applications. They seem to work well with Asterisk, and are easy to configure. Don't count on fax working at all though, or even worse working in some cases... AudioCodes is where we finally found a product that does what we need. They are about twice as much as a Grandstream (at least for the MP-124 vs GXW-4024) but have been rock solid for faxing so far. They also come in multiple configurations which is handy. We use the MP-114 2FXS/2FXO device at our remote sites for local PSTN access and to connect a fax machine. They also support survivability (proxy registration) in case of WAN failure. The complaints that I have are that the web interface has A LOT going on, and there is no real CLI to speak of. Neither of these are real issues, just takes you a few more minutes up front to read the manual. I haven't tried any Adtran devices but have thought about purchasing one to test with if I ever get the time. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS - TDM400 - No dial tone
Ironhide*CLI dialplan show phones
[ Context 'phones' created by 'pbx_config' ]
Include ='internal'[pbx_config]
-= 0 extensions (0 priorities) in 1 context. =-
Ironhide*CLI dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
'1000' = 1. Verbose(1|Extension 1000) [pbx_config]
2. Dial(SIP/1000|30) [pbx_config]
3. Hangup() [pbx_config]
'1001' = 1. Verbose(1|Unrouted call handler) [pbx_config]
2. Answer() [pbx_config]
3. Wait(1)[pbx_config]
4. Playback(our-business-hours-are) [pbx_config]
5. Hangup() [pbx_config]
'500' = 1. Verbose(1|Echo test application) [pbx_config]
2. Playback(our-business-hours-are) [pbx_config]
3. Hangup() [pbx_config]
'_XXX' = 1. Verbose(1|Dial Digicel)[pbx_config]
2. Dial(${LOCAL_OUT_TRUNK}/${EXTEN}) [pbx_config]
3. Congestion() [pbx_config]
4. Hangup() [pbx_config]
-= 4 extensions (15 priorities) in 1 context. =-
Message: 11
Date: Sun, 14 Jun 2009 06:32:16 +0300
From: Tzafrir Cohen tzafrir.co...@xorcom.com
Subject: Re: [asterisk-users] FXS - TDM400 - No dial tone
To: asterisk-users@lists.digium.com
Message-ID: 20090614033216.gf3...@xorcom.com
Content-Type: text/plain; charset=iso-8859-1
On Sat, Jun 13, 2009 at 10:13:11PM -0400, Richard McNeilly wrote:
More Troubleshooting
Ironhide*CLI zap show channels
?? Chan Extension? Context Language?? MOH Interpret
?pseudo??? phones default
? 2??? phones default
? 3??? incoming?? default
? 4??? incoming?? default
dialplan show phones
On Sat, Jun 13, 2009 at 10:13 PM, Richard McNeillyramcnei...@gmail.com wrote:
I have a TMD400 card installed in a PC with one fxs (installed in slot
2) and two fxos (installed in slots 3 4). fxos work fine but I am
unable to get a dial tone for any devices connected to the fxs. I
have correctly connected the power supply to the card and I have even
tried moving the card from slot 1 to 2 on the board.
Below is from the console when I try to route a call from FXO on slot
4 to the FXS on slot 2. Notice the FXS is ringing
-- Executing [...@incoming:1] Verbose(Zap/4-1, 1|Unrouted call
handler) in new stack
Unrouted call handler
-- Executing [...@incoming:2] Answer(Zap/4-1, ) in new stack
-- Executing [...@incoming:3] Wait(Zap/4-1, 1) in new stack
-- Executing [...@incoming:4] Dial(Zap/4-1, Zap/2|30) in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
zaptel.conf
fxoks=2
fxsks=3-4
loadzone=us
defaultzone=us
zapata.conf
[trunkgroups]
;define any trunk groups
[channels]
; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
; define incoming channels
signalling=fxs_ks ; Use FXS signaling for an FXO channel
context=incoming
callerid=asreceived
group=1
channel = 3,4 ; PSTN attached to port 3
; define outgoing channels
signalling=fxo_ks ; Use FXS signaling for an FXO channel
context=phones
group=2
channel = 2
More Troubleshooting
Ironhide*CLI zap show channels
Chan Extension Context Language MOH Interpret
pseudo phones default
2 phones default
3 incoming default
4 incoming default
ztcfg -vvv
Zaptel Version: 1.4.11
Echo Canceller: OSLEC
Configuration
==
Channel map:
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
3 channels to configure.
Software-wise things seem ok and I am certain that I have the power
connected to the PCI card correctly.
Any suggestions as to what I may be doing wrong here?
Richard
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Re: [asterisk-users] FXS - TDM400 - No dial tone
On Sat, Jun 13, 2009 at 10:13:11PM -0400, Richard McNeilly wrote: More Troubleshooting Ironhide*CLI zap show channels Chan Extension Context Language MOH Interpret pseudo phones default 2 phones default 3 incoming default 4 incoming default dialplan show phones -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
There is indeed... well i was about to say there was, but it turns out the one i've got is an fxo adapter, have a look and see if sangoma have any fxs adapters in the series, it seems to be called the usbfxo u100 2009/5/26 Diogo Saad diogos...@gmail.com What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
what I want to do is to answers to mobile calls using a regular phone. Is a usb fxs all I need? Does this u100 have smooth integration with Asterisk ? On Tue, May 26, 2009 at 11:55 AM, Geraint Lee gera...@gmail.com wrote: There is indeed... well i was about to say there was, but it turns out the one i've got is an fxo adapter, have a look and see if sangoma have any fxs adapters in the series, it seems to be called the usbfxo u100 2009/5/26 Diogo Saad diogos...@gmail.com What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
How about a low cost ATA? Just plug the ATA into the network, configure it - along with a SIP definition within sip.conf and you're ready to go. Lee From: Diogo Saad [mailto:diogos...@gmail.com] Sent: Tuesday, May 26, 2009 10:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FXS What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
Using an ATA, Do I still need a softphone or it´s embedded in the hardware? On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
how do I configure my SIP account information? I mean, sip proxy and etc. On Tue, May 26, 2009 at 1:19 PM, Jon Pounder j...@inline.net wrote: Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
Diogo Saad wrote: how do I configure my SIP account information? I mean, sip proxy and etc. you need just a couple pieces of information server (put this in any setting that says proxy or host etc, all set the same) account (the extension in asterisk, put anywhere that sounds like a non-display only field) password (secret, key, password etc., should be one field that takes this in the config) register = yes basically that's it. you mean need to disable feature codes etc, but the above will get most any sip device working with asterisk once you setup an extension for it. On Tue, May 26, 2009 at 1:19 PM, Jon Pounder j...@inline.net mailto:j...@inline.net wrote: Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org http://asterisk.org@sedwards.com http://sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Line Voltage When Dahdi/Zaptel is off?
You mean when the driver is not loaded ? It doesn't. The driver enables the current drawn. Well that is my guess. But since I have one card handy I'll confirm for you. CONFIRMED. No power without the driver loaded Excellent. Thanks, Martin! I didn't have one to test with (yet). Martin On Thu, Apr 2, 2009 at 4:38 PM, Noah Miller noahisaacmil...@gmail.com wrote: Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Line Voltage When Dahdi/Zaptel is off?
On Thu, Apr 02, 2009 at 11:28:04PM -0500, Martin wrote: You mean when the driver is not loaded ? It doesn't. The driver enables the current drawn. Note: that's when the *driver* is loaded. Regardless of whether or not the channel is configured with Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Line Voltage When Dahdi/Zaptel is off?
You mean when the driver is not loaded ? It doesn't. The driver enables the current drawn. Well that is my guess. But since I have one card handy I'll confirm for you. CONFIRMED. No power without the driver loaded Martin On Thu, Apr 2, 2009 at 4:38 PM, Noah Miller noahisaacmil...@gmail.com wrote: Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Help Needed...
I have used the xorcom usb units for fax a few times, and they work pretty well. PaulH Gregory Malsack wrote: Hello All, I have a need to connect an analog device to an asterisk server. The analog device has 4 analog lines going into it (it’s a fax solution). The fax solution answers the analog call, then listens for dtmf. The dtmf code that is played tells the fax device what email address to send the fax to. All calls on our system come into the server through a PRI. The faxes come in over a PRI, the current phone system routes the faxes to the device, then sends the dtmf, then bridges the fax transmission. Does anyone know how I can do this on an asterisk system? I have the PRI card, and have an 8 port fxs card in the system as well. Is it as easy as picking up the line and dialing the 4 digit dtmf, just like it was an fxo port? Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.552 / Virus Database: 270.10.6/1888 - Release Date: 1/12/2009 7:04 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS Help Needed...
those cards don't terminate faxes directly; that is, they aren't fax modems. You can redirect the call to a fake fax with hylafax and asterisk. 2009/1/12 Gregory Malsack gmals...@gmellc.com: Hello All, I have a need to connect an analog device to an asterisk server. The analog device has 4 analog lines going into it (it's a fax solution). The fax solution answers the analog call, then listens for dtmf. The dtmf code that is played tells the fax device what email address to send the fax to. All calls on our system come into the server through a PRI. The faxes come in over a PRI, the current phone system routes the faxes to the device, then sends the dtmf, then bridges the fax transmission. Does anyone know how I can do this on an asterisk system? I have the PRI card, and have an 8 port fxs card in the system as well. Is it as easy as picking up the line and dialing the 4 digit dtmf, just like it was an fxo port? Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.552 / Virus Database: 270.10.6/1888 - Release Date: 1/12/2009 7:04 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS port doesn't provide dialtone
On Fri, Jun 20, 2008 at 10:15:48AM +0200, Paul Schewietzek wrote: Hello everyone, I want to connect a fax to an FXS port (TDM420P). For testing purposes, I connected an analogue phone to it first. However, when I pick it up, I cannot hear anything at all. Is Asterisk actually running? Configured to use those channels? What is the output of: cat /proc/zaptel/* asterisk -rx 'zap show channels' -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS port doesn't provide dialtone
maggie1:~# cat /proc/zaptel/*
Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) HDB3/CCS/CRC4
IRQ misses: 31
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
Span 2: WCTDM/0 Wildcard TDM410P Board 1
IRQ misses: 5
32 WCTDM/0/0 FXSKS
33 WCTDM/0/1 FXSKS
34 WCTDM/0/2 FXOKS (In use)
35 WCTDM/0/3 FXOKS (In use)
maggie1:~# asterisk -rx 'zap show channels'
Chan Extension Context Language MOH Interpret
pseudofrom-analog de default
1from-pstn de default
2from-pstn de default
3from-pstn de default
4from-pstn de default
5from-pstn de default
6from-pstn de default
7from-pstn de default
8from-pstn de default
9from-pstn de default
10from-pstn de default
11from-pstn de default
12from-pstn de default
13from-pstn de default
14from-pstn de default
15from-pstn de default
17from-pstn de default
18from-pstn de default
19from-pstn de default
20from-pstn de default
21from-pstn de default
22from-pstn de default
23from-pstn de default
24from-pstn de default
25from-pstn de default
26from-pstn de default
27from-pstn de default
28from-pstn de default
29from-pstn de default
30from-pstn de default
31from-pstn de default
34from-faxde default
35from-analog de default
Asterisk is running and working for all SIP phones and the TE121,
connected to an E1 :)
I'm beginning to wonder if the card (the TDM400) is actually OK, or if
the FXS module might be broken...
About the channel configuration: Is it invalid to associate a group to
an analog line? Right now, I said Asterik to Dial(ZAP/g4/${EXTEN}) when
the extension for channel 34 gets called from outside. Could it be
possible that a channel with FXO signalling ignores the group= option in
zapata.conf?
Am Freitag, den 20.06.2008, 11:24 +0300 schrieb Tzafrir Cohen:
On Fri, Jun 20, 2008 at 10:15:48AM +0200, Paul Schewietzek wrote:
Hello everyone,
I want to connect a fax to an FXS port (TDM420P). For testing purposes,
I connected an analogue phone to it first. However, when I pick it up, I
cannot hear anything at all.
Is Asterisk actually running?
Configured to use those channels?
What is the output of:
cat /proc/zaptel/*
asterisk -rx 'zap show channels'
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Re: [asterisk-users] FXS port doesn't provide dialtone
Hi not the issue here, but yo asked and thus I'll answer: On Fri, Jun 20, 2008 at 11:51:27AM +0200, Paul Schewietzek wrote: Could it be possible that a channel with FXO signalling ignores the group= option in zapata.conf? A. no problem with that. B. This is only related to dialing out, not to incoming calls. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS, Power and Sangoma
That does indeed sound a bit odd. I've run 12-48 FXS ports from a single molex connector with Sangoma hardware. Try testing your power supply with a multimeter to ensure its putting out the proper voltage. I would not trust the extnernal AC adapters as I've found they typically have voltage that is too low... Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Todd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 1, 2008 8:20:36 PM (GMT-0600) America/Chicago Subject: [asterisk-users] FXS, Power and Sangoma Hi I've a Sangoma A200D with 2FXO and 2FXS. When using it with only the FXO module, it's all good. But when I put in the FXS module and connect the power, logs tells me not enough power. Mar 31 14:11:54 phone kernel: [ 4761.246931] wanpipe1: Module 1: Failed to powerup within 600 ms (8V : 72V)! Mar 31 14:11:54 phone kernel: [ 4761.246937] wanpipe1: Module 1: Did you remember to plug in the power cable? So I disconnect other power devices in the box (Dell Optiplex GX270) such as the Zip drive and CDROMs, but no luck. Then I took an external power supply with molex connector (http://www.coolerguys.com/840556029977.html or http://www.cablesonline.com/mo4inpotoacp.html) and tried that with still the same thing. How much power does this card need? The AC adapter puts out up to 2 Amps. Sangoma support is only telling me to get a bigger power supply and don't use the AC adapter. Has anyone else seen this issue? Could something else be wrong? thanks Todd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I assume span 2 is set ti T1... Also you should be using a crossover if your going from the card direct to the channel bank. Remember an RJ48 crossover and an RH45 crossover are not the same.. It you are using an RJ48 crossover and your span 2 is T1 then try auto T1 If All else fails you can contact support... Free via IAX and FWD for international look at my signature below for details Lee, John (Sydney) wrote: Any luck with the channel bank? Thanks for the reminder Paul but so far no luck. I have been getting: 1) *** Initialising: Trying to frame D4 / ESF on the channel bank 2) Red flashing light on port 2 of the TE412P card I have checked a few things here and there but I think I must have missed some basic stuff. The funny thing is before I purchase the Rhino channel bank, I have been assured that it will work although we are using E1 downunder. Here is my configuration: Asterisk box TE412P Port 1 --- E1 Port 2 --- Rhino 24 port FXS (CB24-FXS-UNIV) Port 3 Port 4 [zaptel.conf] # # E1 # span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 # # Rhino 24-port Channel Bank # span=2,0,0,esf,b8zs fxols=32-55 Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47e882dd139816087317984! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Have you tried setting the card as being T1 instead of E1 for the port connected to the channel bank? PaulH On Tue, 2008-03-25 at 15:33 +1100, Lee, John (Sydney) wrote: Any luck with the channel bank? Thanks for the reminder Paul but so far no luck. I have been getting: 1) *** Initialising: Trying to frame D4 / ESF on the channel bank 2) Red flashing light on port 2 of the TE412P card I have checked a few things here and there but I think I must have missed some basic stuff. The funny thing is before I purchase the Rhino channel bank, I have been assured that it will work although we are using E1 downunder. Here is my configuration: Asterisk box TE412P Port 1 --- E1 Port 2 --- Rhino 24 port FXS (CB24-FXS-UNIV) Port 3 Port 4 [zaptel.conf] # # E1 # span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 # # Rhino 24-port Channel Bank # span=2,0,0,esf,b8zs fxols=32-55 Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I assume span 2 is set ti T1... Thanks James. I will check. Also you should be using a crossover if your going from the card direct to the channel bank. Remember an RJ48 crossover and an RH45 crossover are not the same.. It you are using an RJ48 crossover and your span 2 is T1 then try auto T1 I hope so. I was using the red cable that comes with the product. Do you by any chance have the pin settings of an RJ48 crossover? I want to make a few by myself as a backup. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Have you tried setting the card as being T1 instead of E1 for the port connected to the channel bank? Thanks Paul. I was thinking about the same thing as I was leaving work today. I will try to set the jumper just on port 2 and let you know. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
on an 8 pin connector (rj48) copper up facing away pins labled left to right 1-8 side a 1 white/blue side a 2 blue/white side a 4 white/orange side a 5 orange/white side b 1 white/orange side b 2 orange/white side b 4 white/blue side b 5 blue/white Lee, John (Sydney) wrote: I assume span 2 is set ti T1... Thanks James. I will check. Also you should be using a crossover if your going from the card direct to the channel bank. Remember an RJ48 crossover and an RH45 crossover are not the same.. It you are using an RJ48 crossover and your span 2 is T1 then try auto T1 I hope so. I was using the red cable that comes with the product. Do you by any chance have the pin settings of an RJ48 crossover? I want to make a few by myself as a backup. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47e8aee2139815237839251! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I was thinking about the same thing as I was leaving work today. I will try to set the jumper just on port 2 and let you know. Yes, that fixed the problem. Thanks James and Paul. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Good to here, I know the time off set US - AU is terrible when you need support. Lee, John (Sydney) wrote: I was thinking about the same thing as I was leaving work today. I will try to set the jumper just on port 2 and let you know. Yes, that fixed the problem. Thanks James and Paul. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1031,47e9ace3139819793911839! -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Good to here, I know the time off set US - AU is terrible when you need support. I have continued to configure the analogue phone by just adding new extensions (just like any VOIP phone) to extensions.conf as follows: exten = 5162,1,SetMusicOnHold(cpwr) exten = 5162,n,Dial(Zap/32,20) exten = 5162,n,VoiceMail,5162 exten = 5162,n,Playback(vm-goodbye) exten = 5162,n,Wait(2) exten = 5162,n,HangUp() I was able to call out and call in. However, I noticed that if I dial from the analog phone to a VOIP phone, asterisk shows up as the dialler on the VOIP phone. This is because it is not registered in SIP. Because I think analog phone does not use SIP, so I thought I don't need to configure sip.conf. Am I correct? Did I miss anything in configuring an analog phone? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I think you can set callerid's in zaptel.conf for each analog port - I did that for a client a while ago. (from memory) PaulH On Wed, 2008-03-26 at 15:53 +1100, Lee, John (Sydney) wrote: Good to here, I know the time off set US - AU is terrible when you need support. I have continued to configure the analogue phone by just adding new extensions (just like any VOIP phone) to extensions.conf as follows: exten = 5162,1,SetMusicOnHold(cpwr) exten = 5162,n,Dial(Zap/32,20) exten = 5162,n,VoiceMail,5162 exten = 5162,n,Playback(vm-goodbye) exten = 5162,n,Wait(2) exten = 5162,n,HangUp() I was able to call out and call in. However, I noticed that if I dial from the analog phone to a VOIP phone, asterisk shows up as the dialler on the VOIP phone. This is because it is not registered in SIP. Because I think analog phone does not use SIP, so I thought I don't need to configure sip.conf. Am I correct? Did I miss anything in configuring an analog phone? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Any luck with the channel bank? PaulH On Wed, 2008-03-19 at 18:09 +1100, Lee, John (Sydney) wrote: What kind of information are you looking for? configuration or? If you look in our manuals our cards and the Digium cards configure the same in zaptel and zapata. Hi James, I have purchased a CB24-FXS-UNIV and am using a TE412P card with a Asterisk 1.4 box. One port of the card is connected to an E1. I was told to connect a second port to the Rhino box. The rest of the procedure is a bit of a mystery to me at this stage but I am ready to dive into it :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Any luck with the channel bank? Thanks for the reminder Paul but so far no luck. I have been getting: 1) *** Initialising: Trying to frame D4 / ESF on the channel bank 2) Red flashing light on port 2 of the TE412P card I have checked a few things here and there but I think I must have missed some basic stuff. The funny thing is before I purchase the Rhino channel bank, I have been assured that it will work although we are using E1 downunder. Here is my configuration: Asterisk box TE412P Port 1 --- E1 Port 2 --- Rhino 24 port FXS (CB24-FXS-UNIV) Port 3 Port 4 [zaptel.conf] # # E1 # span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 # # Rhino 24-port Channel Bank # span=2,0,0,esf,b8zs fxols=32-55 Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
What kind of information are you looking for? configuration or? If you look in our manuals our cards and the Digium cards configure the same in zaptel and zapata. Hi James, I have purchased a CB24-FXS-UNIV and am using a TE412P card with a Asterisk 1.4 box. One port of the card is connected to an E1. I was told to connect a second port to the Rhino box. The rest of the procedure is a bit of a mystery to me at this stage but I am ready to dive into it :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
yeah, boot off a flash card. true. guess use one of the xorcom servers :) You guys should make one with pre-installed ABE for certain larger customers. Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Sat, Mar 08, 2008 at 07:41:50PM +0500, Faraz Khan wrote: Just think of a different alternative: If you consider the cost of a 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it: just put a simple PC at that end of the campus and attach the Astribank to it. A simple PC? Thats just asking for trouble. In my experience if this is an enterprise, you would need atleast a HP ML 150 with redundant PSU and Raid 5 + 1 spare disk, redundant fans EVERYWHERE! :) For such a satelite server? Get a small system with no moving parts. A bit more reliable (and less noisy :-) ) than such a server. Do use a reliable system as your main server. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Sun, Mar 09, 2008 at 10:54:11PM +0500, Faraz Khan wrote: Get a small system with no moving parts. A bit more reliable (and less noisy :-) ) than such a server. Do use a reliable system as your main server. yeah, boot off a flash card. true. guess use one of the xorcom servers :) You guys should make one with pre-installed ABE for certain larger customers. Or, y'know, just go to a dedicated Ethernet FXS port server, which is precisely where we started this conversation. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Fri, Mar 07, 2008 at 08:28:14PM -0500, Steve Totaro wrote: Ethernet/SIP is going to be by far the most flexible. You can have much longer cable runs without some kind of USB repeater device. Switches are cheap, CAT5/6 is cheap. You could put a Quintum Tenor AX 48 Port (for instance) in one section of a building, campus, LAN (WAN if you are daring) and the server could be anywhere, not tied by 15 or 30 foot USB cables. Then if you are doing new wiring, you can run the shortest distance from the location of the SIP FXS device to the phones. Just think of a different alternative: If you consider the cost of a 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it: just put a simple PC at that end of the campus and attach the Astribank to it. You can have redundant, self healing links as well as link aggregation. I cannot see how TDMoE or USB come anywhere close to this flexibility and certainly don't see it being a fit for high port densities like discussed. I see TDM0E as something that a tech guy thought would be cool (and it is but not very practical) and a USB device something suited for the SoHo (but missing the scalability, redundancy, and flexibility that IP gives.) As for USB: this is also what I thought before actually starting to work with it. Sure, there are limitations. But the Linux USB stack is a nice one. As for TDMoE, I know that at least the current ztd-eth in Zaptel is considered broken. Fixing it would be appreciated if you actually want to use it :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
To support the quintum viewpoint we have deployed the Tenor AX 24-Port FXS in mass configurations (200-300 extensions) without issues. In a newer project we are going to do 1000 FXS extensions. They are exceptionally reliable. Steve- I thought Quintum doesnt do 48 Port FXS gateways? Last I found out from quintum was that that max is 48 port FXO or 24 Port FXS. Is this correct? Quoting Steve Totaro [EMAIL PROTECTED]: On Fri, Mar 7, 2008 at 8:02 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote: On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. Is there any reason you'd want to do that on a system of that scale instead of just using Ethernetted FXS boxes on a dedicated 100Base? Even if you didn't want to use reinvite, seems you'd still win just from the less expensive host interface (I can't understand people using T-1 interfaces for FXS channels either, honestly, in the current environment). USB is very cheap. It's in every computer. A dedicated ethernet segment costs more to set up that an extra USB segment (a 10$ for an extra USB controller? 20$ for a USB hub? a bit more for the wiring?). TDMoE is more complicated as the latency is higher and the jitter is larger. Now both thing have been (T1 channel banks, and TDMoE) have been done by others. People do use and buy them. I don't intend to say that they don't. But ours does as well :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Ethernet/SIP is going to be by far the most flexible. You can have much longer cable runs without some kind of USB repeater device. Switches are cheap, CAT5/6 is cheap. You could put a Quintum Tenor AX 48 Port (for instance) in one section of a building, campus, LAN (WAN if you are daring) and the server could be anywhere, not tied by 15 or 30 foot USB cables. Then if you are doing new wiring, you can run the shortest distance from the location of the SIP FXS device to the phones. You can have redundant, self healing links as well as link aggregation. I cannot see how TDMoE or USB come anywhere close to this flexibility and certainly don't see it being a fit for high port densities like discussed. I see TDM0E as something that a tech guy thought would be cool (and it is but not very practical) and a USB device something suited for the SoHo (but missing the scalability, redundancy, and flexibility that IP gives.) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I would however be interested in knowing how these USB channel banks work out in a extremely large environment. Cost/Reliability and management wise.Keep in mind that grandstream now has a 24 port FXS gateway which retails for $700- and their newer 8 port gateways are extremely good. Quoting Steve Totaro [EMAIL PROTECTED]: On Fri, Mar 7, 2008 at 8:02 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote: On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. Is there any reason you'd want to do that on a system of that scale instead of just using Ethernetted FXS boxes on a dedicated 100Base? Even if you didn't want to use reinvite, seems you'd still win just from the less expensive host interface (I can't understand people using T-1 interfaces for FXS channels either, honestly, in the current environment). USB is very cheap. It's in every computer. A dedicated ethernet segment costs more to set up that an extra USB segment (a 10$ for an extra USB controller? 20$ for a USB hub? a bit more for the wiring?). TDMoE is more complicated as the latency is higher and the jitter is larger. Now both thing have been (T1 channel banks, and TDMoE) have been done by others. People do use and buy them. I don't intend to say that they don't. But ours does as well :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Ethernet/SIP is going to be by far the most flexible. You can have much longer cable runs without some kind of USB repeater device. Switches are cheap, CAT5/6 is cheap. You could put a Quintum Tenor AX 48 Port (for instance) in one section of a building, campus, LAN (WAN if you are daring) and the server could be anywhere, not tied by 15 or 30 foot USB cables. Then if you are doing new wiring, you can run the shortest distance from the location of the SIP FXS device to the phones. You can have redundant, self healing links as well as link aggregation. I cannot see how TDMoE or USB come anywhere close to this flexibility and certainly don't see it being a fit for high port densities like discussed. I see TDM0E as something that a tech guy thought would be cool (and it is but not very practical) and a USB device something suited for the SoHo (but missing the scalability, redundancy, and flexibility that IP gives.) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Sun, Mar 9, 2008 at 9:24 AM, Faraz Khan [EMAIL PROTECTED] wrote: Steve- I thought Quintum doesnt do 48 Port FXS gateways? Last I found out from quintum was that that max is 48 port FXO or 24 Port FXS. Is this correct? Yep, just up yo 24 FXS and up to 48 FXO... http://www.quintum.com/enterprise/entspecs.html?id=21 -- Raul Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Just think of a different alternative: If you consider the cost of a 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it: just put a simple PC at that end of the campus and attach the Astribank to it. A simple PC? Thats just asking for trouble. In my experience if this is an enterprise, you would need atleast a HP ML 150 with redundant PSU and Raid 5 + 1 spare disk, redundant fans EVERYWHERE! :) Dont get me wrong. I think the idea of having a USB channel bank is great- but deployment in distributed or a 'campus' network would be very problematic compared to a quintum/grandstream SIP gateway (completely solid state / no pc required/ consumes little power) -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Sat, Mar 08, 2008 at 07:41:50PM +0500, Faraz Khan wrote: Just think of a different alternative: If you consider the cost of a 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it: just put a simple PC at that end of the campus and attach the Astribank to it. A simple PC? Thats just asking for trouble. In my experience if this is an enterprise, you would need atleast a HP ML 150 with redundant PSU and Raid 5 + 1 spare disk, redundant fans EVERYWHERE! :) For such a satelite server? Get a small system with no moving parts. A bit more reliable (and less noisy :-) ) than such a server. Do use a reliable system as your main server. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Why not get a TDMoE multiplexer, check out http://spidermux.com/ On Sat, Mar 8, 2008 at 4:03 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Mar 08, 2008 at 07:41:50PM +0500, Faraz Khan wrote: Just think of a different alternative: If you consider the cost of a 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it: just put a simple PC at that end of the campus and attach the Astribank to it. A simple PC? Thats just asking for trouble. In my experience if this is an enterprise, you would need atleast a HP ML 150 with redundant PSU and Raid 5 + 1 spare disk, redundant fans EVERYWHERE! :) For such a satelite server? Get a small system with no moving parts. A bit more reliable (and less noisy :-) ) than such a server. Do use a reliable system as your main server. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Sat, Mar 8, 2008 at 11:35 AM, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Why not get a TDMoE multiplexer, check out http://spidermux.com/ Maybe I am missing the concept, but why would you get a TDMoE multiplexer for the OP's usage? I can't really think under what circumstances this would be valuable. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. Is there any reason you'd want to do that on a system of that scale instead of just using Ethernetted FXS boxes on a dedicated 100Base? Even if you didn't want to use reinvite, seems you'd still win just from the less expensive host interface (I can't understand people using T-1 interfaces for FXS channels either, honestly, in the current environment). Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Fri, Mar 7, 2008 at 3:00 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. Is there any reason you'd want to do that on a system of that scale instead of just using Ethernetted FXS boxes on a dedicated 100Base? Even if you didn't want to use reinvite, seems you'd still win just from the less expensive host interface (I can't understand people using T-1 interfaces for FXS channels either, honestly, in the current environment). Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) http://www.quintum.com/enterprise/en_productdetail.html?id=19 48 Port, you cannot go wrong although, you are going to pay a bit, well worth it. Besides, breaking it down per port makes it a little more palatable. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Fri, Mar 7, 2008 at 5:43 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Mar 7, 2008 at 3:00 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. Is there any reason you'd want to do that on a system of that scale instead of just using Ethernetted FXS boxes on a dedicated 100Base? Even if you didn't want to use reinvite, seems you'd still win just from the less expensive host interface (I can't understand people using T-1 interfaces for FXS channels either, honestly, in the current environment). Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) http://www.quintum.com/enterprise/en_productdetail.html?id=19 48 Port, you cannot go wrong although, you are going to pay a bit, well worth it. Besides, breaking it down per port makes it a little more palatable. Thanks, Steve Totaro Sorry, wrong link. http://www.quintum.com/enterprise/en_productdetail.html?id=21 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
200 extensions, take 100 PAP2 and you´re set. The trouble would be configuring them all. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote: On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. Is there any reason you'd want to do that on a system of that scale instead of just using Ethernetted FXS boxes on a dedicated 100Base? Even if you didn't want to use reinvite, seems you'd still win just from the less expensive host interface (I can't understand people using T-1 interfaces for FXS channels either, honestly, in the current environment). USB is very cheap. It's in every computer. A dedicated ethernet segment costs more to set up that an extra USB segment (a 10$ for an extra USB controller? 20$ for a USB hub? a bit more for the wiring?). TDMoE is more complicated as the latency is higher and the jitter is larger. Now both thing have been (T1 channel banks, and TDMoE) have been done by others. People do use and buy them. I don't intend to say that they don't. But ours does as well :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Fri, Mar 7, 2008 at 8:02 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote: On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. Is there any reason you'd want to do that on a system of that scale instead of just using Ethernetted FXS boxes on a dedicated 100Base? Even if you didn't want to use reinvite, seems you'd still win just from the less expensive host interface (I can't understand people using T-1 interfaces for FXS channels either, honestly, in the current environment). USB is very cheap. It's in every computer. A dedicated ethernet segment costs more to set up that an extra USB segment (a 10$ for an extra USB controller? 20$ for a USB hub? a bit more for the wiring?). TDMoE is more complicated as the latency is higher and the jitter is larger. Now both thing have been (T1 channel banks, and TDMoE) have been done by others. People do use and buy them. I don't intend to say that they don't. But ours does as well :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Ethernet/SIP is going to be by far the most flexible. You can have much longer cable runs without some kind of USB repeater device. Switches are cheap, CAT5/6 is cheap. You could put a Quintum Tenor AX 48 Port (for instance) in one section of a building, campus, LAN (WAN if you are daring) and the server could be anywhere, not tied by 15 or 30 foot USB cables. Then if you are doing new wiring, you can run the shortest distance from the location of the SIP FXS device to the phones. You can have redundant, self healing links as well as link aggregation. I cannot see how TDMoE or USB come anywhere close to this flexibility and certainly don't see it being a fit for high port densities like discussed. I see TDM0E as something that a tech guy thought would be cool (and it is but not very practical) and a USB device something suited for the SoHo (but missing the scalability, redundancy, and flexibility that IP gives.) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
Chris Bagnall wrote: Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel banks on reasonably sizeable installs I'd be interested to hear what device(s) you used, how simple or complex they were to configure, and whether there'd be any issues attaching multiple units to a single server. This install would be in the UK, so we do need to factor in the different conditions expected by UK POTS handsets (line impedance, etc.). Are most channel banks country-neutral, or do specific models need to be purchased for different line conditions in each country? Thanks in advance. Regards, Chris www.citel.com I used them a few years back in a pilot install with legacy Nortel phones and it worked well. I gather they have grown tremendously from there. I'm in North America, don't know how well they support UK stuff. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Thu, Mar 06, 2008 at 03:21:47PM -, Chris Bagnall wrote: I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel banks on reasonably sizeable installs I'd be interested to hear what device(s) you used, how simple or complex they were to configure, and whether there'd be any issues attaching multiple units to a single server. This install would be in the UK, so we do need to factor in the different conditions expected by UK POTS handsets (line impedance, etc.). Are most channel banks country-neutral, or do specific models need to be purchased for different line conditions in each country? You might want to check the archives from, I think, early '07; I was looking into doing a hotel/motel system for a client, and asked almost exactly this question. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Thu, Mar 06, 2008 at 10:38:36AM -0500, Drew Gibson wrote: www.citel.com I used them a few years back in a pilot install with legacy Nortel phones and it worked well. I gather they have grown tremendously from there. I'm in North America, don't know how well they support UK stuff. Citel are, are they not, the company that specializes in FXS channel banks specific to legacy digital phones? Do they do analog-POTS banks as well? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Thu, Mar 6, 2008 at 10:49 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Mar 06, 2008 at 10:38:36AM -0500, Drew Gibson wrote: www.citel.com I used them a few years back in a pilot install with legacy Nortel phones and it worked well. I gather they have grown tremendously from there. I'm in North America, don't know how well they support UK stuff. Citel are, are they not, the company that specializes in FXS channel banks specific to legacy digital phones? Do they do analog-POTS banks as well? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) Citel is the worst product I have ever dealt with, worse than Grandstream but for different reasons. Anyways, for smaller port density I love the Quintum Tenor AX 24 port FXS, They may make a 48, I am not sure. This is a SIP connection, and there are probably a multitude of other products that do the same, Quintum blew me away with the sheer amount of options and configuration (that you will probably never use). I have heard people suggest MaxTNT for high port densities, which looks great, I just have no experience or need for such a device yet. The other option is a channel bank that connects via T1 or I guess E1 (although I have never seen an E1 30 port channel bank, I am in the US so it is not surprising) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I have been told to use Rhino Channel Bank but I am yet to set it up and I appreciate if someone can show me some doco of using Rhino on an E1/T1 with TE410. Thanks. I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What kind of information are you looking for? configuration or? If you look in our manuals our cards and the Digium cards configure the same in zaptel and zapata. Lee, John (Sydney) wrote: I have been told to use Rhino Channel Bank but I am yet to set it up and I appreciate if someone can show me some doco of using Rhino on an E1/T1 with TE410. Thanks. I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47d0790d14234975420232! - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, Come with a Money Back gurantee and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH0HucdloC7YyaIOoRAv5zAJ9jdZQEkXbYfbvP7QbONR+DVVYSdQCfSmmb dv00H0l/fgJiTU9o4Z6++9Y= =wDLO -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. I know I've missed the original message in this thread, so it'll be a bit out of place, but what about the Xorcom Channel banks? e.g.: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? However, even with E1 units, you're still looking at 7 E1 ports... (2 quad cards + the external channel bank) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel banks
On Thu, Mar 06, 2008 at 11:50:43PM +, Gordon Henderson wrote: I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. I know I've missed the original message in this thread, so it'll be a bit out of place, but what about the Xorcom Channel banks? e.g.: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS damaged at TDM22B
I'd say check with Digium, maybe it's supposed to not break (i personally don't think it would break it, i'd have noticed it already :) if you plug it to the wrong thing and you will get a replacement for free. Zoa bilal ghayyad wrote: Hi All; If one of my FXS port damaged at TDM22B because we connected the Telephone Line cable to the FXS port while it should be connected to the FXO port, then can I order S110M FXS Module and fix it instead of the damaged FXS? (This if we assume my problem that really the FXS port damaged). Rregards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS damaged at TDM22B
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Zoa wrote: I'd say check with Digium, maybe it's supposed to not break (i personally don't think it would break it, i'd have noticed it already :) if you plug it to the wrong thing and you will get a replacement for free. I thought on the older TDM400P cards that the voltage coming in from a ring would destroy an FXS module? We have a strange situation here in New Zealand where you have analogue DDI numbers. Basically you provide a dialtone to the exchange, the exchange provides a dialtone to you, and if a normal call comes in it rings you as normal. If a DDI call comes in, it seizes the line, then dials the extension (last four digits of the DDI). I was running a site off a GXW4008 gateway, but it had problems (I'm assuming with the bidirectional ringer voltages). We changed them over to BRI :) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHlUTuDQNt8rg0Kp4RAgvQAJ0XQMsuWMFsBZ58LCo1DVqfJQa8fgCeKg9H NPtBeFkfAc6DshBZe52aGqg= =ebva -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel bank
On Jun 28, 2007, at 8:00 AM, pixiesfr wrote: hello, We looking for a channel bank to connect 120 analogs phones, in SIP to an Asterisk .. Did someone have some product in mind. A channel bank must connect via a T1 by definition, which would then give you 24 phone lines per T1. This would require 5 T1 connected to your asterisk server. OK 4 if E1 as it probably is in your case. However with your requirement for SIP you are looking for a gateway to connect your phones. Most are 24 port, though some are 48 port. Names to look at would be Carrier Access, Audiocodes, Vega etc. I do like the Vega unit except for their support - or lack thereof - here in the US. They do have both 24 and 48 port units. Your other option would be to do GR303 which would allow you to hang many lines off a few T1/E1 circuits, except it is definately not SIP. If phones are not at location of your asterisk server and you really want to do sip, it may be simpler for this many phones to install an additional asterisk server at the remote location and install a quad port T1/E1 card and hang channel banks off it. Good Luck ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS channel bank
On Thu, 28 Jun 2007, Jerry Jones wrote: Your other option would be to do GR303 which would allow you to hang many lines off a few T1/E1 circuits, except it is definately not SIP. Well, and it's probably worth pointing out that if you wanted to go the GR.303 route, the devices on both ends -- and especially the service provider side -- is unlikely to be so inexpensive and simple as a mere Asterisk server. You'd need some sort of DLC capable of doing GR.303, and, well, I don't know what supports GR.303 subscriber interfaces on the service side other than a bona fide Class 5 switch of some sort. :-) -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS + Pots Extensions Help
We wanted a cheap last resort fail-over. A few really cheap pots lines are easy to run buy, as we can get them for a really low cost. My understanding with DIDs (and its limited), is they have to belong to a PRI. The only way that is cheaper than a few pots lines is if you needed 8 or more pots lines. Then the line fees balance out. I was hoping for a solution more along the lines of Use this x variable that contains what ZAP channel it came in on, then I can program that one to point to a particular person. Thanks, Rob Sean M. Pappalardo wrote: Rob Schall wrote: Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the s extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? Sure, it's called a DID trunk. It's basically just a regular analog phone line but the CO switch sends down the digits dialed in one of a few ways: Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency (DTMF). They are usually inbound-only, but some CO's can add outbound service too if needed. Call your phone service provider's business office and ask about analog DID lines/trunks. They should be around $30/mo for the line and $1-2/mo for each number. Ask them what type of signaling they use then you'll need to configure your zapata.conf to match. After that, you can then start routing in the dialplan based on the number called. For extra fun, have the CO set them up in a hunt group to avoid busy signals. Take a look at: http://www.voip-info.org/wiki-Asterisk+tips+DID (BTW, Why are you adding analog lines if you're already big enough for a PRI? Isn't it less expensive to just add a couple more DID numbers to the PRI?) Sean - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, Inc. - ___ --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] FXS + Pots Extensions Help
Rob Schall wrote: My understanding with DIDs (and its limited), is they have to belong to a PRI. DID can be delivered over a PRI, a channelized T1 or over analog trunks. If you use the analog route method, you can get any number of trunks. The only way that is cheaper than a few pots lines is if you needed 8 or more pots lines. Then the line fees balance out. I was hoping for a solution more along the lines of Use this x variable that contains what ZAP channel it came in on, then I can program that one to point to a particular person. If you are only really interested in distinguishing which pots line the call came in on and want to make decisions based on that then you will just put each line in a different context in zapata.conf. Then your dial plan will dial phone1 for context1, phone2 for context2 and so on. Don Pobanz ___ --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] FXS + Pots Extensions Help
Alex Balashov wrote: Sure, it's called a DID trunk. It's basically just a regular analog phone line but the CO switch sends down the digits dialed in one of a Sean, I am curious--what do these look like these days? Are they ordinary T1s? CAS/robbed-bit? Do these just use the signaling portions associated with each channel to deliver the winks, and do the channels correspond to the appropriate timeslots on the voice trunk? How does this work? I'm not sure of the technical details as I don't have one installed yet, but from my discussions with the phone companies and some phone installers I know, these are just your average analog tip/ring lines. They just have some extra signaling to get the digits to you. Give a call to your phone service provider and ask to speak with a technical group. Sean - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, Inc. - ___ --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] FXS + Pots Extensions Help
Here's a silly question, if these are standard POTS you obviously know which number corresponds to which line, being the case couldn't you tell that ZAP/1 is POTS 555-1234, ZAP/2 is POTS 555-1235, etc etc? I'm assuming you're trying to identify the inbound number from the telco that was dialed. Unless you have the lines in a hunt group at the telco, but then you're implying you don't care which number was dialed, you just want failover at the telco. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Wednesday, May 23, 2007 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FXS + Pots Extensions Help We wanted a cheap last resort fail-over. A few really cheap pots lines are easy to run buy, as we can get them for a really low cost. My understanding with DIDs (and its limited), is they have to belong to a PRI. The only way that is cheaper than a few pots lines is if you needed 8 or more pots lines. Then the line fees balance out. I was hoping for a solution more along the lines of Use this x variable that contains what ZAP channel it came in on, then I can program that one to point to a particular person. Thanks, Rob Sean M. Pappalardo wrote: Rob Schall wrote: Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the s extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? Sure, it's called a DID trunk. It's basically just a regular analog phone line but the CO switch sends down the digits dialed in one of a few ways: Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency (DTMF). They are usually inbound-only, but some CO's can add outbound service too if needed. Call your phone service provider's business office and ask about analog DID lines/trunks. They should be around $30/mo for the line and $1-2/mo for each number. Ask them what type of signaling they use then you'll need to configure your zapata.conf to match. After that, you can then start routing in the dialplan based on the number called. For extra fun, have the CO set them up in a hunt group to avoid busy signals. Take a look at: http://www.voip-info.org/wiki-Asterisk+tips+DID (BTW, Why are you adding analog lines if you're already big enough for a PRI? Isn't it less expensive to just add a couple more DID numbers to the PRI?) Sean - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, Inc. - ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] FXS + Pots Extensions Help
Rob Schall wrote: understanding with DIDs (and its limited), is they have to belong to a PRI. The only way that is cheaper than a few pots lines is if you needed This is not true, since analog DID trunks do exist. I was hoping for a solution more along the lines of Use this x variable that contains what ZAP channel it came in on, then I can program that one to point to a particular person. Actually with DID, you'd just program the dialplan the same as for a PRI, since the variables for DID should be the same regardless of which channel type/technology the call comes in on. (You wouldn't tie Zap FXO channels directly to extensions unless using non-DID POTS lines where each has only one number, which doesn't sound like what you are wanting to do.) Sean - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, Inc. - ___ --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] FXS + Pots Extensions Help
On Wed, 23 May 2007, Sean M. Pappalardo said something to this effect: Give a call to your phone service provider and ask to speak with a technical group. I do not share your optimism about the revelation this would entail. :-) But thank you! -- Alex -- Alex Balashov [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] FXS + Pots Extensions Help
Jeremy,
This is the best thing i was able to come up with.
All incoming pots lines go to the zapchans context
[zapchans]
exten = 3,1,Dial(ZAP/1-1);ZAP3
exten = 3,2,Hangup()
exten = 4,1,Dial(ZAP/2-1);ZAP4
exten = 4,2,Hangup()
exten = s,1,Answer()
exten = s,2,Goto(${CHANNEL:4:1},1)
exten = s,3,Hangup()
Each one could have its own context, but I wanted to keep it all in one
place and make it easy for our phone guys to handle (they aren't linux
or asterisk saavy).
The only problem I've found so far, is when you dial in from a pots
line, the call connects fine, but they won't hang each other up. The
console shows the hangup command running, but neither side of the call
will hangup when the opposite side hangs their line up. Not sure if I
just missed a setting or what.
Jeremy Mann wrote:
Here's a silly question, if these are standard POTS you obviously know which
number corresponds to which line, being the case couldn't you tell that ZAP/1
is POTS 555-1234, ZAP/2 is POTS 555-1235, etc etc?
I'm assuming you're trying to identify the inbound number from the telco that
was dialed. Unless you have the lines in a hunt group at the telco, but then
you're implying you don't care which number was dialed, you just want
failover at the telco.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Wednesday, May 23, 2007 8:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FXS + Pots Extensions Help
We wanted a cheap last resort fail-over. A few really cheap pots lines
are easy to run buy, as we can get them for a really low cost. My
understanding with DIDs (and its limited), is they have to belong to a
PRI. The only way that is cheaper than a few pots lines is if you needed
8 or more pots lines. Then the line fees balance out.
I was hoping for a solution more along the lines of Use this x
variable that contains what ZAP channel it came in on, then I can
program that one to point to a particular person.
Thanks,
Rob
Sean M. Pappalardo wrote:
Rob Schall wrote:
Normally I just use pri's with our asterisk systems, but a request came
in to add some normal pots lines to the setup. We have 3 lines, and they
run into the fxs ports. They hit the dialplan just fine, and they always
dial the s extension. However, my question would be... Is there a way
to determine what number was dialed and have it forward to a specific
phone?
Sure, it's called a DID trunk. It's basically just a regular analog
phone line but the CO switch sends down the digits dialed in one of a
few ways: Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency
(DTMF). They are usually inbound-only, but some CO's can add outbound
service too if needed. Call your phone service provider's business
office and ask about analog DID lines/trunks. They should be around
$30/mo for the line and $1-2/mo for each number. Ask them what type of
signaling they use then you'll need to configure your zapata.conf to
match. After that, you can then start routing in the dialplan based on
the number called. For extra fun, have the CO set them up in a hunt
group to avoid busy signals.
Take a look at: http://www.voip-info.org/wiki-Asterisk+tips+DID
(BTW, Why are you adding analog lines if you're already big enough for
a PRI? Isn't it less expensive to just add a couple more DID numbers
to the PRI?)
Sean
-
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please notify Texas Health Management
Re: [asterisk-users] FXS + Pots Extensions Help
On Tue, 22 May 2007, Sean M. Pappalardo said something to this effect: Sure, it's called a DID trunk. It's basically just a regular analog phone line but the CO switch sends down the digits dialed in one of a few ways: Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency (DTMF). They are usually inbound-only, but some CO's can add outbound service too if needed. Sean, I am curious--what do these look like these days? Are they ordinary T1s? CAS/robbed-bit? Do these just use the signaling portions associated with each channel to deliver the winks, and do the channels correspond to the appropriate timeslots on the voice trunk? How does this work? -- Alex Balashov [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] FXS + Pots Extensions Help
Rob Schall wrote: Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the s extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? Sure, it's called a DID trunk. It's basically just a regular analog phone line but the CO switch sends down the digits dialed in one of a few ways: Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency (DTMF). They are usually inbound-only, but some CO's can add outbound service too if needed. Call your phone service provider's business office and ask about analog DID lines/trunks. They should be around $30/mo for the line and $1-2/mo for each number. Ask them what type of signaling they use then you'll need to configure your zapata.conf to match. After that, you can then start routing in the dialplan based on the number called. For extra fun, have the CO set them up in a hunt group to avoid busy signals. Take a look at: http://www.voip-info.org/wiki-Asterisk+tips+DID (BTW, Why are you adding analog lines if you're already big enough for a PRI? Isn't it less expensive to just add a couple more DID numbers to the PRI?) Sean - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, Inc. - ___ --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] FXS gateway/Channel Bank
Adtran TA750 or TA850 Roger Workman wrote: Can someone recommend a good FXS gateway/Channel bank that will intergrate smoothly with * I need to port over 158 analog lines -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] FXS adapters and Polycom phones
hi, i would prefer a Mediant 1000 with 12 ports FXS of Audiocodes to do the Job. further info available upon request, Mickey On 7/12/06, Mike [EMAIL PROTECTED] wrote: Hi, I`m looking for a SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but resuing their own Norstar PSTN phones. They have 10phones. From a price point of view, it seems that 10 individual GrandStream SIP adapters is the best way to go, but it seems so inelegant to me. What is recommended ? Second question: I have a GrandStream GXP-2000, that despite what everybody says I love. I am still looking for a replacement, if only because it doesn`t look as good and it does have a few quirks. I was looking at Polycoms, but some questions are unanswered by looking at their datasheet. - Does the Polycom 501 have an integrated router (like the GXP-2000, latest firmware, does) - Can you have more than one SIP/account on the phone, each ringing in a way that lets the user know which account is ringing? (GXP2000 does it by making it possible to have each line linked to a separate SIP account) Thank you, Mike___--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] FXS adapters and Polycom phones
Mike wrote: I`m looking for a SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but resuing their own Norstar PSTN phones. They have 10 phones. From a price point of view, it seems that 10 individual GrandStream SIP adapters is the best way to go, but it seems so inelegant to me. What is recommended ? Not sure about the others, but I've had decent experiences with the Linksys PAP2 series, and they aren't that expensive. Second question: I have a GrandStream GXP-2000, that despite what everybody says I love. I am still looking for a replacement, if only because it doesn`t look as good and it does have a few quirks. I was looking at Polycoms, but some questions are unanswered by looking at their datasheet. - Does the Polycom 501 have an integrated router (like the GXP-2000, latest firmware, does) Not entirely sure what you're asking here. If you're wondering if it has a two NIC interfaces (a pass-through for the PC) then yes. - Can you have more than one SIP/account on the phone, each ringing in a way that lets the user know which account is ringing? (GXP2000 does it by making it possible to have each line linked to a separate SIP account) The 501 is capable of having 3 different line appearances, each of which can have a primary and secondary server configured for them. -- Jamin W. Collins ___ --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] FXS: No ringtone
On Jul 10, 2006, at 1:23 AM, yusuf wrote: Hi all, I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before Oh yeah, what you are talking about is ring back, not ringtone. I think the r option in the asterisk dial command might help you as that forces ringback. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS: No ringtone
Martin Joseph wrote: On Jul 10, 2006, at 1:23 AM, yusuf wrote: Hi all, I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before Oh yeah, what you are talking about is ring back, not ringtone. I think the r option in the asterisk dial command might help you as that forces ringback. The r option seldom fixes ringback issues. Make sure you have /etc/asterisk/indications.conf setup. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] FXS Caller ID revisted
Are you doing something funny with the CID on it's way to the phone? I've got a somewhat similar problem with an Aastra IP phone (yes, I did say IP): it would NOT ring if the caller id started with an #. Maybe your Aastra PSTN phone got some of the same (buggy?) handling of CID's? Dan Elder wrote: Hi All, posted last week but didn't get any responses. I'm trying to figure out why some of our analog phones aren't showing CID when hooked up to asterisk. To recap, I have an Aastra Powertouch 350, which shows caller ID fine when connected to the PSTN, but when hooked up to asterisk, CID does not show. I've hooked up another phone to the same * port that the Aastra phone is on, it DOES show CID, so I'm assuming my settings such are at least partially correct, can anyone point me to some options or areas I can look to troubleshoot this issue? Been pulling my hair out on this for days just can't seem to get it sorted. I'm using asterisk 1.2.0 with a Carrier Access ABII channel bank. When another CID capable phone is hooke up to the same port, CID works fine, the Aastra phone is however unable to read the incoming CID from * apparently. Any pointers would be greatly appreciated, I've searched the Wiki the CID faq's to no avail. Thanks in advance Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] FXS channel banks
as of now, 8 ports for 8 phones. there will be soon a 16 ports version (within April) On 3/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: How many phones lines ? -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] De la part de Curt ShafferEnvoyé: vendredi 24 mars 2006 03:17À: asterisk-users@lists.digium.com Objet: [Asterisk-Users] FXS channel banks Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt ___--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] FXS channel banks
Title: Message As of now we are probably looking in the 36 range. We would like to utilize this as a first step to migrating to a VoIP system. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 25, 2006 2:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] FXS channel banks How many phones lines ? -Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Curt Shaffer Envoyé: vendredi 24 mars 2006 03:17 À: asterisk-users@lists.digium.com Objet: [Asterisk-Users] FXS channel banks Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt smime.p7s Description: S/MIME cryptographic signature ___ --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] FXS channel banks
Title: Message Curt Shaffer wrote: As of now we are probably looking in the 36 range. We would like to utilize this as a first step to migrating to a VoIP system. To save cost. I would buy a couple of used Adtran 750's, they are cheap and readily available for under $500 with FXS cards. Combined with a Dual T1 card, you have an inexpensive solution, and when you go SIP, you can sell the channel banks for what you paid for them. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] FXS channel banks
smime.p7m Description: S/MIME encrypted message ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : RE : [Asterisk-Users] FXS channel banks
2 x TDM2460E (with hardware echocan module) or TDM2460B (wo/echocan) = 48 phones lines, no T1 cards, no channel banks level adjustments troubles, direct Zap channels and simple switching. Probably the best choice and price :-) Best Regards, Francois BERGERET, France. A very happy TDM2400 user ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE : RE : [Asterisk-Users] FXS channel banks
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, March 25, 2006 12:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : RE : [Asterisk-Users] FXS channel banks 2 x TDM2460E (with hardware echocan module) or TDM2460B (wo/echocan) = 48 phones lines, no T1 cards, no channel banks level adjustments troubles, direct Zap channels and simple switching. Probably the best choice and price :-) Best Regards, Francois BERGERET, France. A very happy TDM2400 user ;-) I love Quintum TenorAX boxes. You can put the box anywhere and point back to your Asterisk box. It has so many features it is really an amazing box and support is excellent. Thanks, Steve Totaro ___ --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] FXS channel banks
i would suggest Astribank-8 of XorCom. it is a dedicated Asterisk compliant solution. you can look at : www.xorcom.com. On 3/24/06, Curt Shaffer [EMAIL PROTECTED] wrote: Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt___--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] FXS channel banks
Tele Cost Price Reducer wrote: i would suggest Astribank-8 of XorCom. it is a dedicated Asterisk compliant solution. you can look at : www.xorcom.com. Looks interesting, shame they don't have a FXO version. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] FXS channel banks
Title: Message How many phones lines ? -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Curt ShafferEnvoyé: vendredi 24 mars 2006 03:17À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] FXS channel banks Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt ___ --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] FXS channel banks
Carrier Access Adit 600 On 3/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt ___ --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] FXS channel banks
rhino channel bankOn 3/23/06, C F [EMAIL PROTECTED] wrote: Carrier Access Adit 600On 3/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelito Manansalawww.voicefidelity.netMobile: +63 906 437 0459DID: (+63) 44 7906770US DID: +1 619 399 0128 msn: [EMAIL PROTECTED]skype: bulcrack ___ --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] FXS with v.90 modem support?
Thanks for the info! Fortunately, we have something closer to the latter configuration you described. The PSTN goes directly into an MX250 as a SIP gateway, and our Asterisk server connects to that. The MX has a few FXO ports, but we don't want to use them. It doesn't seem very clean to have an IP phone system but yet still needing to run analog lines throughout the building. Although, the Atlas 550 card would be nice if our Asterisk box were a little closer to where the analog devices are (and if they have OS X/PPC drivers for it). Has anyone ever used the Vega 50? That seems the most promising solution so far. - .Dustin On Feb 6, 2006, at 6:21 PM, Steve Underwood wrote: Dustin Wenz wrote: I have a couple of devices that need an analog modem to communicate outside of our Asterisk system. Most FXS gateways don't seem to support this... I have a stack of Sipura 2002's that are, AFAIK, worthless for this purpose. I've heard that Digium's IAXy FXS will work with modems, but I can't find any reference to that in their documentation. There is also the VegaStream Vega 50 that claims to support v.90 on 8 FXS ports per unit. Does anyone have experience with these devices, or can recommend anything else? Then important things here are that you must only digitise once, and that the path must be clean and free of timing slips. Things like modem - FXS - asterisk - FXO - PSTN will not work. The signal is digitised, returned to analogue, then the PSTN will digitise it again. In general VoIP boxes of any kind will not work, as they do not guarantee a clean path. What should work is modem - FXS - asterisk - E1 or t1 - PSTN However, a lot of people have problems getting even this configuration to give clean enough results for V.90 to work. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] FXS with v.90 modem support?
Little expensive, but Adtran Atlas 550 + Octal analog card works fine. $5k. -Original Message- From: Dustin Wenz [mailto:[EMAIL PROTECTED] Sent: Monday, February 06, 2006 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FXS with v.90 modem support? I have a couple of devices that need an analog modem to communicate outside of our Asterisk system. Most FXS gateways don't seem to support this... I have a stack of Sipura 2002's that are, AFAIK, worthless for this purpose. I've heard that Digium's IAXy FXS will work with modems, but I can't find any reference to that in their documentation. There is also the VegaStream Vega 50 that claims to support v.90 on 8 FXS ports per unit. Does anyone have experience with these devices, or can recommend anything else? - .Dustin Wenz ___ --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
