[asterisk-users] Channel Bank ? Simple Switch Hangup?
I am trying to configure a channel bank with 24 ports of FXS., but appear to
be hitting a roadblock? This worked on v1.4.xx but now just get
SimpleSwitch and immediate=no/yes don't seem to make a difference?, no
matter if under top section, under channel, etc.
Chan_dahdi.conf:
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A104 port 3 [slot:1 bus:1 span:3] wanpipe3
context=from-cb
group=3
echocancel=yes
signalling=fxo_ls
channel = 49-72
immediate=yes
Extensions.conf:
[from-cb]
exten = s,1,DISA,no-password|internal
[internal]
include = sip-stations
include = iax-trunks
include = outbound
[outbound]
exten = _1XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XXX,1,Dial(DAHDI/g1/${EXTEN})
When I pickup a line, and hit any key I get:
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/72-1'
-- Hungup 'DAHDI/72-1'
Asterisk Version 1.6.2.13
Lastest DAHDI/LibPRI/SpanDSP
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Re: [asterisk-users] Channel Bank ? Simple Switch Hangup?
Nevermind, figured it out.
Immediate=yes on top part of chan_dahdi.conf
And in extensions.conf
Exten =s,1,disa(no-password,internal)
William Stillwell
Systems Architect
MDT Personnel, LLC.
Ph. Coming soon.
Fx. Coming soon.
Cl. 727-638-6208
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell (Lists)
Sent: Tuesday, October 26, 2010 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Channel Bank ? Simple Switch Hangup?
I am trying to configure a channel bank with 24 ports of FXS., but appear to
be hitting a roadblock? This worked on v1.4.xx but now just get
SimpleSwitch and immediate=no/yes don't seem to make a difference?, no
matter if under top section, under channel, etc.
Chan_dahdi.conf:
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A104 port 3 [slot:1 bus:1 span:3] wanpipe3
context=from-cb
group=3
echocancel=yes
signalling=fxo_ls
channel = 49-72
immediate=yes
Extensions.conf:
[from-cb]
exten = s,1,DISA,no-password|internal
[internal]
include = sip-stations
include = iax-trunks
include = outbound
[outbound]
exten = _1XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XXX,1,Dial(DAHDI/g1/${EXTEN})
When I pickup a line, and hit any key I get:
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/72-1'
-- Hungup 'DAHDI/72-1'
Asterisk Version 1.6.2.13
Lastest DAHDI/LibPRI/SpanDSP
--
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Re: [asterisk-users] Channel Bank ? Simple Switch Hangup?
Have you contacted Sangoma regarding their card configuration?
I have found them always very knowledgeable and helpful
I would certainly go there first.
John Novack
William Stillwell (Lists) wrote:
I am trying to configure a channel bank with 24 ports of FXS., but
appear to be hitting a roadblock? This worked on v1.4.xx but now just
get SimpleSwitch and immediate=no/yes don't seem to make a
difference?, no matter if under top section, under channel, etc.
Chan_dahdi.conf:
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A104 port 3 [slot:1 bus:1 span:3] wanpipe3
context=from-cb
group=3
echocancel=yes
signalling=fxo_ls
channel = 49-72
immediate=yes
Extensions.conf:
[from-cb]
exten = s,1,DISA,no-password|internal
[internal]
include = sip-stations
include = iax-trunks
include = outbound
[outbound]
exten = _1XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XXX,1,Dial(DAHDI/g1/${EXTEN})
When I pickup a line, and hit any key I get:
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/72-1'
-- Hungup 'DAHDI/72-1'
Asterisk Version 1.6.2.13
Lastest DAHDI/LibPRI/SpanDSP
--
Dog is my Co-pilot
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[asterisk-users] Channel Bank Recommendations
Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank Thanks Mark ___ --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] Channel Bank Recommendations
On Wed, 2007-08-29 at 09:38 -0400, Mark Bell wrote: Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank Personally, I've had great success with Carrier Access (ADIT 600) and Adtran (TA-750/TA-850) channel banks (even the ones I've bought at bargain prices on eBay). Just avoid the Carrier Access AccessBank 1 models with FXO ports... the FXO ports have no disconnect supervision. (They're the ones that are shaped like a pizza box.) -Jared ___ --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] Channel Bank Recommendations
On 8/29/07, Mark Bell [EMAIL PROTECTED] wrote: Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank We're using a Rhino here and haven't had one problem with it. It's connected to an analog fax server and lights up for hours at a time. Probably been up 300+ days since we bought 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] Channel Bank Recommendations
Mark Bell wrote: Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank I have an Adtran Total Access 750 on my system, and it has worked very well. However, if you live near a radio transmitter, you will need RF filters for your FXO ports. I live about three miles from a 50,000-watt AM transmitter, and without filters, my FXOs pick up the radio station nicely. :) Commonly-available DSL filters will work. Russ ___ --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] Channel Bank Recommendations
Good to know thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Russ Price Sent: Wednesday, August 29, 2007 8:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Channel Bank Recommendations Mark Bell wrote: Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank I have an Adtran Total Access 750 on my system, and it has worked very well. However, if you live near a radio transmitter, you will need RF filters for your FXO ports. I live about three miles from a 50,000-watt AM transmitter, and without filters, my FXOs pick up the radio station nicely. :) Commonly-available DSL filters will work. Russ ___ --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] Channel Bank
On Sunday 06 May 2007 6:42 pm, Forum wrote: Can someone recommend a good quality 24 or greater port channel bank? For FXS: I have personally used Adit600, Access Bank I and IIs. They all work great, and the AB1 and AB2 products are *cheap*. For FXO: Adit600. The AB1/2 work, but have no CPD capability, so you can never tell when the other side hangs up on you. Rhino makes 'em too, as does Adtran, but I have no experience with these. -A. ___ --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] Channel Bank
I'm a fan of Adtran Total Access (TA) 750s. They are so cheap on eBay that you can get two of them used for less than the cost of a new one. Follow the standard things for not getting ripped off on eBay, of course. Andrew Kohlsmith wrote: On Sunday 06 May 2007 6:42 pm, Forum wrote: Can someone recommend a good quality 24 or greater port channel bank? For FXS: I have personally used Adit600, Access Bank I and IIs. They all work great, and the AB1 and AB2 products are *cheap*. For FXO: Adit600. The AB1/2 work, but have no CPD capability, so you can never tell when the other side hangs up on you. Rhino makes 'em too, as does Adtran, but I have no experience with these. -A. ___ --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] Channel Bank
Eric ManxPower Wieling wrote: I'm a fan of Adtran Total Access (TA) 750s. They are so cheap on eBay that you can get two of them used for less than the cost of a new one. Follow the standard things for not getting ripped off on eBay, of course. The TA750s are nice indeed - they're built like tanks, and they can be configured with all sorts of different interfaces. However, be aware that the FXO cards are sensitive to RF interference. Standard DSL line filters will eliminate the interference, though. I live about three miles from a 50 kW AM transmitter, and the filters are vital. Aside from eBay, there are used-equipment vendors that provide refurbished cards and such at reasonable prices. I got my FXO card that way when I couldn't find any on eBay. FXOs are a bit expensive, though, and that goes for eBay or vendors. On FXS ports, avoid using the automatic line impedance setting. In my experience, it would eventually guess wrong; the phones would work fine for a while, then randomly become incredibly loud. On checking, I discovered that the TA750 was picking 900 ohm+2.16uF complex impedance instead of plain 900 ohm. Once the FXS ports were forced to 900 ohm, there were no further problems. ___ --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] Channel Bank
Russ Price wrote: Eric ManxPower Wieling wrote: I'm a fan of Adtran Total Access (TA) 750s. They are so cheap on eBay that you can get two of them used for less than the cost of a new one. Follow the standard things for not getting ripped off on eBay, of course. The TA750s are nice indeed - they're built like tanks, and they can be configured with all sorts of different interfaces. However, be aware that the FXO cards are sensitive to RF interference. Standard DSL line filters will eliminate the interference, though. I live about three miles from a 50 kW AM transmitter, and the filters are vital. I'll have to try that. I have hum on one of my TA750s and suspected a hardware problem, but I've got DSL filters laying around to try. On the other hand I live 11 miles from the CO on a SLIC96 about 3 miles away. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel Bank
Can someone recommend a good quality 24 or greater port channel bank? 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
Re: [asterisk-users] Channel Bank
Forum wrote: Can someone recommend a good quality 24 or greater port channel bank? The Adit 600 is a favorite of mine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel Bank
On 5/6/07, Doug Lytle [EMAIL PROTECTED] wrote: The Adit 600 is a favorite of mine. Doug Would second the Adit too. We are running a Rhino now and have had no problems with it. -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel bank log
Hello Can any one may send me log when channel bank is work Best regards Viktor Tatianin ___ --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] Channel bank not work
Hello Bill Zapata.conf switchtype=national secallerid=yes echocancel = yes echocancelwhenbridged = yes rxgain = 0.0 txgain = 0.0 signalling=fxo_ks immediate=no #include zapata_additional.conf context=from-internal group=1 channel=110 *CLI -- Starting simple switch on 'Zap/110-1' -- Hungup 'Zap/110-1' *CLI But at handset is silence :-( May any one send config for channel bank Please -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Schaffer Sent: Wednesday, July 05, 2006 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work In your /etc/asterisk/zapata.conf file, what do you have the immediate keyword set to? I think it needs to be set to no if you want dialtone and digit collection. Also, changes in this file require a full stop and restart of asterisk. -Bill On Mon, Jul 03, 2006 at 08:05:58PM +0300, Viktor Tatianin wrote: After ring I hangup phone but don't speak at the phone silence :- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work OK so * is seeing the phone go offhook That is good How about if you call the handset, can you actually talk across the connection? You said it rang, but now we need to establish you actually have an audio connection. On Jul 3, 2006, at 9:50 AM, Viktor Tatianin wrote: If lift up handset 3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on 'Zap/94-1' Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous match)... Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Hangup: channel: 94 index = 0, normal = 42, callwait = -1, thirdcall = -1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: disabled echo cancellation on channel 94 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/94-1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Updated conferencing on 94, with 0 conference users Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank not work
Hello When I lift handset at phone hear silence This is my config *CLI zap show channel 110 Channel: 110 File Descriptor: 42 Span: 4 Extension: Dialing: no Context: from-internal Caller ID: 2812 Calling TON: 0 Caller ID name: 2812 Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook *CLI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work OK so * is seeing the phone go offhook That is good How about if you call the handset, can you actually talk across the connection? You said it rang, but now we need to establish you actually have an audio connection. On Jul 3, 2006, at 9:50 AM, Viktor Tatianin wrote: If lift up handset 3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on 'Zap/94-1' Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous match)... Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Hangup: channel: 94 index = 0, normal = 42, callwait = -1, thirdcall = -1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: disabled echo cancellation on channel 94 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/94-1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Updated conferencing on 94, with 0 conference users Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Channel bank not work
On Fri, Jul 07, 2006 at 06:48:07PM +0300, Viktor Tatianin wrote: Hello Bill Zapata.conf switchtype=national secallerid=yes echocancel = yes echocancelwhenbridged = yes rxgain = 0.0 txgain = 0.0 signalling=fxo_ks immediate=no #include zapata_additional.conf What do you have in zapata_additional.conf ? Those three are irrelevant: context=from-internal group=1 channel=110 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank not work
All I can offer you is blank looks. The only practical difference I see between your setup excerpts and mine is that I am using loopstart and you are using kewlstart. Kewlstart allows for signalling that the far end has disconnected the call. I think this should show up on your console trace, though. I don't recall seeing it. Maybe you are looking at the wrong end of the problem? Are you sure the channel bank is working as expected? -Bill On Fri, Jul 07, 2006 at 06:53:54PM +0300, Viktor Tatianin wrote: Hello When I lift handset at phone hear silence This is my config *CLI zap show channel 110 Channel: 110 File Descriptor: 42 Span: 4 Extension: Dialing: no Context: from-internal Caller ID: 2812 Calling TON: 0 Caller ID name: 2812 Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook *CLI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work OK so * is seeing the phone go offhook That is good How about if you call the handset, can you actually talk across the connection? You said it rang, but now we need to establish you actually have an audio connection. On Jul 3, 2006, at 9:50 AM, Viktor Tatianin wrote: If lift up handset 3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on 'Zap/94-1' Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous match)... Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Hangup: channel: 94 index = 0, normal = 42, callwait = -1, thirdcall = -1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: disabled echo cancellation on channel 94 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/94-1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Updated conferencing on 94, with 0 conference users Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Channel bank not work
In your /etc/asterisk/zapata.conf file, what do you have the immediate keyword set to? I think it needs to be set to no if you want dialtone and digit collection. Also, changes in this file require a full stop and restart of asterisk. -Bill On Mon, Jul 03, 2006 at 08:05:58PM +0300, Viktor Tatianin wrote: After ring I hangup phone but don't speak at the phone silence :- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work OK so * is seeing the phone go offhook That is good How about if you call the handset, can you actually talk across the connection? You said it rang, but now we need to establish you actually have an audio connection. On Jul 3, 2006, at 9:50 AM, Viktor Tatianin wrote: If lift up handset 3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on 'Zap/94-1' Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous match)... Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Hangup: channel: 94 index = 0, normal = 42, callwait = -1, thirdcall = -1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: disabled echo cancellation on channel 94 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/94-1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Updated conferencing on 94, with 0 conference users Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Channel bank not work
Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --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] Channel bank not work
Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --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] Channel bank not work
If lift up handset 3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on 'Zap/94-1' Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous match)... Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Hangup: channel: 94 index = 0, normal = 42, callwait = -1, thirdcall = -1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: disabled echo cancellation on channel 94 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/94-1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Updated conferencing on 94, with 0 conference users Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank not work
Show channels asterisk1*CLI show channels Channel Location State Application(Data) Zap/94-1 [EMAIL PROTECTED]:1Rsrvd (None) 1 active channel 0 active calls But at phone is silence -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank not work
OK so * is seeing the phone go offhook That is good How about if you call the handset, can you actually talk across the connection? You said it rang, but now we need to establish you actually have an audio connection. On Jul 3, 2006, at 9:50 AM, Viktor Tatianin wrote: If lift up handset 3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on 'Zap/94-1' Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous match)... Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Hangup: channel: 94 index = 0, normal = 42, callwait = -1, thirdcall = -1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: disabled echo cancellation on channel 94 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/94-1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Updated conferencing on 94, with 0 conference users Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank not work
After ring I hangup phone but don't speak at the phone silence :- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work OK so * is seeing the phone go offhook That is good How about if you call the handset, can you actually talk across the connection? You said it rang, but now we need to establish you actually have an audio connection. On Jul 3, 2006, at 9:50 AM, Viktor Tatianin wrote: If lift up handset 3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on 'Zap/94-1' Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous match)... Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Hangup: channel: 94 index = 0, normal = 42, callwait = -1, thirdcall = -1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: disabled echo cancellation on channel 94 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/94-1 Jul 3 17:49:21 DEBUG[14197] chan_zap.c: Updated conferencing on 94, with 0 conference users Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 03, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank not work Are you seeing any messages on the console? You should be seeing something like Starting simple switch We would need more info to help more. On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote: Hello All Please help me, I have next problem When lift up handset at phone which connect to channel bank I don't hear dialtone but if I dial this number phone ring When after ring hungup handset at phone voice not work Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Channel bank woes - no outbound calls
[internal] exten = 5148346,1,Dial(Zap/g1/514836) Anybody out there have any ideas on why all of the digits aren't being sent out? Shouldn't this be like this ? exten = 5148346,1,Dial(Zap/g1/5148346) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank woes - no outbound calls
So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it goes through just fine. However, if I attempt an outbound call, I get Your call did not go through. Please try your call again. After much experimenting, I found out this happens if you dial some digits, but not enough for a full phone number. My zaptel.conf looks like: span=1,0,0,esf,b8zs fxsks=1 loadzone=us defaultzone=us And my zapata.conf looks like: [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=no transfer=no echocancel=no ; define channels group=1 context=from-pstn signalling=fxs_ks channel = 1 And finally, extensions.conf looks like: [from-pstn] exten = 6080,1,Answer() exten = 6080,2,Playback(hello-world) exten = 6080,3.Hangup() [internal] exten = 5148346,1,Dial(Zap/g1/514836) Anybody out there have any ideas on why all of the digits aren't being sent out? Thanks, James ___ --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] Channel bank woes - no outbound calls
[EMAIL PROTECTED] wrote: So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it goes through just fine. However, if I attempt an outbound call, I get Your call did not go through. Please try your call again. After much experimenting, I found out this happens if you James, When I have problems with outbound on my Adit 600, it's usually because I have signaling screwed up on that channel. (i.e. trying to grab a groundstart line with loopstart signaling). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank woes - no outbound calls
I'm hooked up to a regular analog POTS line. I've tried both loop start and ground start, but no luck either way. Any other thoughts? Thanks, James Doug Lytle wrote: [EMAIL PROTECTED] wrote: So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it goes through just fine. However, if I attempt an outbound call, I get Your call did not go through. Please try your call again. After much experimenting, I found out this happens if you James, When I have problems with outbound on my Adit 600, it's usually because I have signaling screwed up on that channel. (i.e. trying to grab a groundstart line with loopstart signaling). Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank woes - no outbound calls
James Texter wrote: I'm hooked up to a regular analog POTS line. I've tried both loop start and ground start, but no luck either way. Any other thoughts? Unfortunately, no. I only have experience with the Adit 600 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank timing
On 12/26/05 08:28 Andrew Kohlsmith said the following: There are two problems with this: 1. the A104 can have each span's sync independent of the others, unlike the Digium cards. 2. With both spans trying to sync to each other you can run into interesting clock situations you may want to avoid. what would the equivalent be for the digium cards ? would something like the following work ? span=1,0,0 span=2,1,0 span=3,2,0 span=4,0,0 (note that span's 1 and 4 are set as PRI NET) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Channel bank timing
On Tuesday 27 December 2005 05:25, Dinesh Nair wrote: what would the equivalent be for the digium cards ? would something like the following work ? span=1,0,0 span=2,1,0 span=3,2,0 span=4,0,0 (note that span's 1 and 4 are set as PRI NET) What is each span connected to? Remember that Digium multispan cards can only have one clock 'master' for all spans. Span 3 in this case is shown as the secondary source, which means that the device connected to span 3 is expecting to be the clocking source. This means that there will be frame slips and other nasties (HDLC aborts for PRI, etc.) on span 3 until span 2 goes down and span 3 becomes the sync source. -A. ___ --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] Channel bank timing
Can you get just one channel bank working? What exactly does it sound like? Frame slips sound like the occassional "chirp" or buzz. I have always had one working. It was adding the second that caused so much trouble. It sounds like dropouts in the speech, short little dropouts.. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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] Channel bank timing
I don't believe the above config is correct. It should have been fine. Both channel banks will be generating timing/clock signals within their transmit leg towards the asterisk box. That is part of T1/E1 low level protocol design and you can't change it even if you wanted to. Yes, but both channel banks can sync to the line, and the Sangoma card can be set to not sync to the line, thus becoming the master on both spans. On the asterisk T1 port connected to CB2, use: span=2,2,0,esf,b8zs where the second 2 tells your asterisk T1 card to use this port for sync if the first port does dead, fails, cable is disconnected, or for any other reason that would essentially represent a failure of CB1. There are two problems with this: 1. the A104 can have each span's sync independent of the others, unlike the Digium cards. 2. With both spans trying to sync to each other you can run into interesting clock situations you may want to avoid. Ops, wasn't aware each span had independent clock/syncing. Sorry. ___ --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] Channel bank timing
On Monday 26 December 2005 07:20, Chris Mason (Lists) wrote: I have always had one working. It was adding the second that caused so much trouble. It sounds like dropouts in the speech, short little dropouts.. Do you have trouble on *both* when you add the second? What happens if you swap the ports the that channel banks plug in to? Does the problem stick with the span or the channel bank? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank timing
Can anyone help me understand channel bank timing? I have a server with a Sangoma A104 T1 card connected to two channel banks and I am having audio problems that is clearly timing errors. I thought I understood how to configure it but clearly I don't. All my incoming lines are PSTN, I do not have access to PRI. All my extension phones are SIP. My asterisk version is 1.2.1. Channel bank 1: Adtran 600 with 12 FXO: Timing set to Network on CB /etc/zapata.conf: span=1,0,0,esf,b8zs Channel Bank 2: Adtran 750 with 12 FXO Timing set to loop on channel bank controller span=2,0,0,esf,b8zs With this configuration I am getting choppy sound. What should they be set to? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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] Channel bank timing
Can anyone help me understand channel bank timing? I have a server with a Sangoma A104 T1 card connected to two channel banks and I am having audio problems that is clearly timing errors. I thought I understood how to configure it but clearly I don't. All my incoming lines are PSTN, I do not have access to PRI. All my extension phones are SIP. My asterisk version is 1.2.1. Channel bank 1: Adtran 600 with 12 FXO: Timing set to Network on CB /etc/zapata.conf: span=1,0,0,esf,b8zs Channel Bank 2: Adtran 750 with 12 FXO Timing set to loop on channel bank controller span=2,0,0,esf,b8zs With this configuration I am getting choppy sound. What should they be set to? I don't believe the above config is correct. Pick one of the channel banks and declare it as your source for timing. I'll pick CB1 so as to follow through the words below. Both channel banks will be generating timing/clock signals within their transmit leg towards the asterisk box. That is part of T1/E1 low level protocol design and you can't change it even if you wanted to. On the asterisk T1 port connected to CB1, use: span=1,1,0,esf,b8zs where the second 1 tells your asterisk T1 card to use this port for syncing the onboard T1 clock (on the Sangoma card). On the asterisk T1 port connected to CB2, use: span=2,2,0,esf,b8zs where the second 2 tells your asterisk T1 card to use this port for sync if the first port does dead, fails, cable is disconnected, or for any other reason that would essentially represent a failure of CB1. On CB2, configure it to obtain its clock sync from the T1. (I don't have any Adtrans around, so can't tell you exactly what the setting words are.) On CB1, configure it to use internal T1 clocking (whatever words those happen to be for an Adtran). If you just want to play around without changing the CB at all, just change the second digit in span=1,0,0,esf,b8zs to indicate that its your source for timing. Only one span= statement can have a 1. I'm not 100% sure on this next statement, but would guess you'll need to stop asterisk and reload the T1 card drivers (or simply reboot). ___ --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] Channel bank timing
On Sunday 25 December 2005 18:39, Chris Mason (Lists) wrote: Can anyone help me understand channel bank timing? I have a server with a Sangoma A104 T1 card connected to two channel banks and I am having audio problems that is clearly timing errors. I thought I understood how to configure it but clearly I don't. All my incoming lines are PSTN, I do not have access to PRI. All my extension phones are SIP. My asterisk version is 1.2.1. Tell the Channel Banks clock to the line, and have the Sangoma card NOT sync to anything (i.e. the A104 is the master, the channel banks the slaves). Basically clocking works this way: Each end of a T1 sends data generated by an on-board clock. These two clocks (one at each side) needs to be in perfect sync with each other or you get frame slips and other nasties. The solution is to have one of these clocks lock or synchronize to the far side. This is know by several names, among them line clock, recovered clock, slave clock, etc. The side that is not trying to synchronize is also known my several names... master clock, internal clock, etc. So it comes down to this: One side must synchronize its clock to the other side (which does NOT do this) or you will have frame slips. The Digium multi-span T1/E1 cards can only slave to one clock for the whole card, whereas the Sangoma cards are a little different and can have each span slave to its own clock. For your particular application it doesn't matter. Have the card provide sync (be the master, internal, etc.) and have the channel banks recover clock (be the slave, use line clock, etc.) from the line. Hopefully that helps. :-) Channel bank 1: Adtran 600 with 12 FXO: Timing set to Network on CB /etc/zapata.conf: span=1,0,0,esf,b8zs Channel Bank 2: Adtran 750 with 12 FXO Timing set to loop on channel bank controller span=2,0,0,esf,b8zs That looks perfect. Make sure that you really are setting that. IIRC the Sangoma card needs to have its clock set with the wancfg utility and not just ztcfg. -A. ___ --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] Channel bank timing
On Sunday 25 December 2005 18:47, Rich Adamson wrote: I don't believe the above config is correct. It should have been fine. Both channel banks will be generating timing/clock signals within their transmit leg towards the asterisk box. That is part of T1/E1 low level protocol design and you can't change it even if you wanted to. Yes, but both channel banks can sync to the line, and the Sangoma card can be set to not sync to the line, thus becoming the master on both spans. On the asterisk T1 port connected to CB2, use: span=2,2,0,esf,b8zs where the second 2 tells your asterisk T1 card to use this port for sync if the first port does dead, fails, cable is disconnected, or for any other reason that would essentially represent a failure of CB1. There are two problems with this: 1. the A104 can have each span's sync independent of the others, unlike the Digium cards. 2. With both spans trying to sync to each other you can run into interesting clock situations you may want to avoid. -A. ___ --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] Channel bank timing
Andrew Kohlsmith wrote: Tell the Channel Banks clock to the line, and have the Sangoma card NOT sync to anything (i.e. the A104 is the master, the channel banks the slaves). I set the card up so that Port1 TE_CLOCK= MASTER TE_REF_CLOCK= 0 Port2 TE_CLOCK = NORMAL TE_REF_CLOCK = 1 which should make Port 2 take it's timing from Port 1 and Port 1 take it's timing from the onboard clock. Basically clocking works this way: Each end of a T1 sends data generated by an on-board clock. These two clocks (one at each side) needs to be in perfect sync with each other or you get frame slips and other nasties. The solution is to have one of these clocks lock or synchronize to the far side. This is know by several names, among them line clock, recovered clock, slave clock, etc. The side that is not trying to synchronize is also known my several names... master clock, internal clock, etc. On the 600 I set it to Timing = Network, but on the 750 I can't figure out which one of these it should be. LOOP LOCAL EXTERNAL On the 600, the manual says: The selected clock option always designates the clock source for transmission. Clocking necessary for receiving data is always recovered from incoming data. I think the 600 manual also gives me the answer for the 750: Network Timing - The network is the source of timing. The received data clocking is looped back to the network, where it is used to determine the transmission timing. This option is also referred to as loop timed as the transmission clock is derived from the received clock. So for the 750, loop would be the same thing. So, as far as I can tell, everything is set correctly. Which is a problem because it does not sound right. ___ --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] Channel bank timing
On Sunday 25 December 2005 20:47, Chris Mason wrote: I set the card up so that Port1 TE_CLOCK= MASTER TE_REF_CLOCK= 0 Port2 TE_CLOCK = NORMAL TE_REF_CLOCK = 1 which should make Port 2 take it's timing from Port 1 and Port 1 take it's timing from the onboard clock. Ok; I'm not *that* familliar with the Sangoma cards (I do love their S518 ADSL card though!) so I'll have to believe you on that setup. :-) On the 600 I set it to Timing = Network, but on the 750 I can't figure out which one of these it should be. LOOP LOCAL EXTERNAL Loop. On the 600, the manual says: The selected clock option always designates the clock source for transmission. Clocking necessary for receiving data is always recovered from incoming data. Yup. You want Loop or Network. So for the 750, loop would be the same thing. Correct. So, as far as I can tell, everything is set correctly. Which is a problem because it does not sound right. Can you get just one channel bank working? What exactly does it sound like? Frame slips sound like the occassional chirp or buzz. -A. ___ --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] Channel Bank Help Please....
Sounds like the two devices are not agreeing on the signalling. double check the channel bank settings against the zaptel ones. You did run ztcfg after changing the zaptel file right? zttool should be helpful here. On Aug 2, 2005, at 1:11 PM, David Sampson wrote: Hello – I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, Dave zaptel.conf loadzone = us defaultzone=us span=1,1,0,esf,b8zs fxoks=1-24 zapata.conf [channels] group=1 language=en signalling=fxo_ks usecallerid=no context=default echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=1.0 txgain=1.0 channel = 1-24 extensions.conf exten = 3500,1,Dial,Zap/1|60 ; exten = 3500,2,Hangup exten = 3501,1,Dial,Zap/2|60 ; exten = 3501,2,Hangup exten = 3502,1,Dial,Zap/3|60 ; exten = 3502,2,Hangup exten = 3503,1,Dial,Zap/4|60 ; exten = 3503,2,Hangup exten = 3504,1,Dial,Zap/5|60 ; exten = 3504,2,Hangup exten = 3505,1,Dial,Zap/6|60 ; exten = 3505,2,Hangup exten = 3506,1,Dial,Zap/7|60 ; exten = 3506,2,Hangup exten = 3507,1,Dial,Zap/8|60 ; exten = 3507,2,Hangup exten = 3508,1,Dial,Zap/9|60 ; exten = 3508,2,Hangup exten = 3509,1,Dial,Zap/10|60 ; exten = 3509,2,Hangup When I attempt to call these extensions I get: *CLI dial 3501 -- Executing Dial(OSS/dsp, Zap/2|60) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 1: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 3: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 4: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 5: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 6: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 7: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 8: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 9: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 10: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 11: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 12: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 13: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 14: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 15: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 16: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 17: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 18: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 19: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 20: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 21: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 22: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 23: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 24: Yellow Alarm Aug 2 12:59:50 WARNING[3401]: chan_zap.c:3195 zt_handle_event: Detected alarm on channel 2: Yellow Alarm -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (local, 3501, 2) exited non-zero on 'OSS/dsp' Hangup on console Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 1 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 2 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on
Re: [Asterisk-Users] Channel Bank Help Please....
Actually Adits are pretty much self configuring for fxs anyway - I can install and only have to set ip info On Aug 2, 2005, at 6:38 PM, Doug Lytle wrote: David Sampson wrote: Hello – I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, David, You need to do more then just plug the channel bank in and expect it to work. You need to configure it. If it's anything like an Adit 600, you need to tell the channel bank how to setup each channel on the cards. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank Help Please....
Hello I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, Dave zaptel.conf loadzone = us defaultzone=us span=1,1,0,esf,b8zs fxoks=1-24 zapata.conf [channels] group=1 language=en signalling=fxo_ks usecallerid=no context=default echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=1.0 txgain=1.0 channel = 1-24 extensions.conf exten = 3500,1,Dial,Zap/1|60 ; exten = 3500,2,Hangup exten = 3501,1,Dial,Zap/2|60 ; exten = 3501,2,Hangup exten = 3502,1,Dial,Zap/3|60 ; exten = 3502,2,Hangup exten = 3503,1,Dial,Zap/4|60 ; exten = 3503,2,Hangup exten = 3504,1,Dial,Zap/5|60 ; exten = 3504,2,Hangup exten = 3505,1,Dial,Zap/6|60 ; exten = 3505,2,Hangup exten = 3506,1,Dial,Zap/7|60 ; exten = 3506,2,Hangup exten = 3507,1,Dial,Zap/8|60 ; exten = 3507,2,Hangup exten = 3508,1,Dial,Zap/9|60 ; exten = 3508,2,Hangup exten = 3509,1,Dial,Zap/10|60 ; exten = 3509,2,Hangup When I attempt to call these extensions I get: *CLI dial 3501 -- Executing Dial(OSS/dsp, Zap/2|60) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 1: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 3: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 4: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 5: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 6: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 7: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 8: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 9: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 10: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 11: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 12: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 13: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 14: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 15: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 16: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 17: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 18: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 19: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 20: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 21: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 22: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 23: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 24: Yellow Alarm Aug 2 12:59:50 WARNING[3401]: chan_zap.c:3195 zt_handle_event: Detected alarm on channel 2: Yellow Alarm -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (local, 3501, 2) exited non-zero on 'OSS/dsp' Hangup on console Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 1 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 2 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 3 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 4 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 5 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 6 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 7 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 8 Aug 2
Re: [Asterisk-Users] Channel Bank Help Please....
David Sampson wrote: Hello – I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, David, You need to do more then just plug the channel bank in and expect it to work. You need to configure it. If it's anything like an Adit 600, you need to tell the channel bank how to setup each channel on the cards. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank of E1s? (one E1 input -- 2 x E1 output)
Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank of E1s? (one E1 input -- 2 x E1 output)
yes, some multiplexer allows that, but they're quite expensive compared to another E1 card for asterisk. I think you'll need at least 1k $$$ for a such splitter. Matteo. Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank of E1s? (one E1 input -- 2 x E1 output)
Matteo Brancaleoni wrote: yes, some multiplexer allows that, but they're quite expensive compared to another E1 card for asterisk. I think you'll need at least 1k $$$ for a such splitter. Matteo, would you have any reference for this 'mux/splitter' ? I would guess it need to be smart enough to dig into the signalling, since is not only the PCM DS0s that would need to be Y-splitted. [], O-O Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank of E1s? (one E1 input -- 2 x E1 output)
I just called this company. They seem to do what is required. Now remains the pricing part of it. I will wait for their feedback. http://www.megatelindustries.com/products.htm Hakem, Selon Julio Arruda [EMAIL PROTECTED]: Matteo Brancaleoni wrote: yes, some multiplexer allows that, but they're quite expensive compared to another E1 card for asterisk. I think you'll need at least 1k $$$ for a such splitter. Matteo, would you have any reference for this 'mux/splitter' ? I would guess it need to be smart enough to dig into the signalling, since is not only the PCM DS0s that would need to be Y-splitted. [], O-O Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank of E1s? (one E1 input -- 2 x E1 output)
Many channel banks have two T-1 connectors and support a feature called 'drop and insert'. This allows some of the DS0 channels to be cross connected from one T-1 connection to the other. The first T-1 connection can go to the telco or an interface card in a computer, and the second T-1 can go to another channel bank. Some of the channels can be dropped off at the first channel bank while the rest can continue on to the second channel bank. You are asking about E-1 and PBX instead of T-1 and channel bank, but if I understand the 'drop and insert' correctly, and if your hardware supports it, this may work for you. Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank replacement
I enjoy using the Adit 600 with the new FXS cards via the controller T1 interfaces. Works well. I do have concerns with using the CMG card via MGCP. Has anyone done this? How is it working? On Apr 8, 2005, at 12:50 PM, Matt Schulte wrote: Word of warning, get the version 5 or higher FXS cards with the ADIT600, else you will have echo problems. This is just from personal experience. Supposedly the 5 and higher cards have dynamic impedance adjustment, it's worth it. Matt -Original Message- From: Peter Hoppe [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Channel bank replacement Thank you so much for your answers already, I really appreciate it! I have looked into using an Adtran Total Access 750 platform instead, but got away from that idea after I saw the totally confusing amount of options of different modules I can buy. The Adit 600 seemed so much simpler to put together. Also, the Adit 600 had such an excellent appraisal in the asterisk voip-info - see http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank But maybe I need to come back to the Adtran TA750. Unfortunately that platform seems to only offer 24 fxs ports per unit and I need to buy an expensive T1 card. I would buy the Digium T1 card - it seems that it is by far the least expensive card, but $500 is still something. That's why I toyed with the Adit 600 plus cmg card - all I need is a standard network card on the Asterisk machine. We have sorely abandoned the idea of using an extensive amount of voip phones on the property, as we are not a homogenous office setup (ppl also live on the property). This solution would mean * putting in an entire new cat5 network. I would be the person who would have to put it all in place - When would I be finished? In 2 years? 4 years? 10 years? * lots of admin hassle to enable all the phones / add new phones / remove phones * users can't easily extend stations at end points. With two wire phone they simply switch one parallel to the existing one - no admin hassle / extra hubs etc. * two wire technology enables us to buy almost any phone available. * security concerns with the SIP protocol. See http://secunia.com/advisories/8169/ as an example * users potentially plugging their laptops into the voip sockets and browsing/downloading away = lots of setup/admin hassle with the firewall (how do you block Kazaa?) * Phones potentially breaking when users unplug power during firmware download. For example, this is an issue with the Grandstream phone. The only alternative that seems feasible at the moment would be * a different channel bank than the adit 600 or * a voip gateway that multiplexes many fxs ports into one ethernet connection. But before I would go down that route I would have to be absolutely sure that the SIP conforms to the standard, the upgrades are free and the fxs ports are compatible with uk standard two wire phones. I found that some two wire phones actually use 4 wires - confusing * a bank of ATAs (handytone 286 or similar). I *really* don't like that solution, as it is a bad botch job and throws lots of issues like which REN they have, many power supplies (or one big one). I really ought to be red in the face for even mentioning that solution. But if nothing else is available, I would probably have to buy them in bulk, take the boards out and mount them in a 19'' box together with a hub so I build my own voip gateway :) maybe it's not so botch after all :) ) For connection to the PSTN: We have three BT lines, and again, we would not like to move over to a different technology like ISDN. The lines work for us, and 'if it ain't broke, don't fix it'. We would use three Sipura SPA-3000 interfaces to connect them to the internal network. The SPA-3000 is sold in the UK and has the CE approval, so it should legally be ok. I am experimenting with one unit at the moment, and am smacked by the literally hundreds of options it has. But I heard good reports about that one, so I expect it to work well in our setting. Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. I got an Adtran 600 with 12 X FXO and 12 X FXS cards for $495 from Penny Doyen [EMAIL PROTECTED] With the strength of the pound
[Asterisk-Users] Channel bank replacement
Hello, I am working for a charity in the UK and I am projecting a new phone system. We would like to connect our two-wire telephones (40 or so) to an ADIT 600 channel bank, and connect that into an Asterisk box via the CMG card or T1 card. I have been in talks with Carrier Access about the purchase of a new channel bank and we tried to get a minor version of it first for testing with the intention of upgrading to the full product if we are happy with it. Unfortunately since a few months I cannot get any further with CAC, as they keep not coming back to us on how we proceed. I feel that the channel bank would be the best solution, but it seems that we are just to small fish to fry for them. So - would there be any other way to connect 40+ telephones (two wire) into an asterisk box? Are there any voip gateways that actually conform to SIP standard (unlike what I heard from the Mediatrix voip gateways 1124 and 1204 which seem to use non standard SIP and have pay-as-you-upgrade)? Thank you very much for your consideration! Peter Hoppe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank replacement
Maybe following options: 1-) Get another channel bank from ebay at low cost. Which will also need another T1 card; 2-) Use 40 voip phones at 50 USD each and you no longer need the card neither the channel bank. But a reliable local network ; Selon Peter Hoppe [EMAIL PROTECTED]: Hello, I am working for a charity in the UK and I am projecting a new phone system. We would like to connect our two-wire telephones (40 or so) to an ADIT 600 channel bank, and connect that into an Asterisk box via the CMG card or T1 card. I have been in talks with Carrier Access about the purchase of a new channel bank and we tried to get a minor version of it first for testing with the intention of upgrading to the full product if we are happy with it. Unfortunately since a few months I cannot get any further with CAC, as they keep not coming back to us on how we proceed. I feel that the channel bank would be the best solution, but it seems that we are just to small fish to fry for them. So - would there be any other way to connect 40+ telephones (two wire) into an asterisk box? Are there any voip gateways that actually conform to SIP standard (unlike what I heard from the Mediatrix voip gateways 1124 and 1204 which seem to use non standard SIP and have pay-as-you-upgrade)? Thank you very much for your consideration! Peter Hoppe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank replacement
On Friday 08 April 2005 16:35, Peter Hoppe wrote: Hello, I am working for a charity in the UK and I am projecting a new phone system. So - would there be any other way to connect 40+ telephones (two wire) into an asterisk box? Are there any voip gateways that actually conform to SIP standard (unlike what I heard from the Mediatrix voip gateways 1124 and 1204 which seem to use non standard SIP and have pay-as-you-upgrade)? Thank you very much for your consideration! Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank replacement
Thank you so much for your answers already, I really appreciate it! I have looked into using an Adtran Total Access 750 platform instead, but got away from that idea after I saw the totally confusing amount of options of different modules I can buy. The Adit 600 seemed so much simpler to put together. Also, the Adit 600 had such an excellent appraisal in the asterisk voip-info - see http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank But maybe I need to come back to the Adtran TA750. Unfortunately that platform seems to only offer 24 fxs ports per unit and I need to buy an expensive T1 card. I would buy the Digium T1 card - it seems that it is by far the least expensive card, but $500 is still something. That's why I toyed with the Adit 600 plus cmg card - all I need is a standard network card on the Asterisk machine. We have sorely abandoned the idea of using an extensive amount of voip phones on the property, as we are not a homogenous office setup (ppl also live on the property). This solution would mean * putting in an entire new cat5 network. I would be the person who would have to put it all in place - When would I be finished? In 2 years? 4 years? 10 years? * lots of admin hassle to enable all the phones / add new phones / remove phones * users can't easily extend stations at end points. With two wire phone they simply switch one parallel to the existing one - no admin hassle / extra hubs etc. * two wire technology enables us to buy almost any phone available. * security concerns with the SIP protocol. See http://secunia.com/advisories/8169/ as an example * users potentially plugging their laptops into the voip sockets and browsing/downloading away = lots of setup/admin hassle with the firewall (how do you block Kazaa?) * Phones potentially breaking when users unplug power during firmware download. For example, this is an issue with the Grandstream phone. The only alternative that seems feasible at the moment would be * a different channel bank than the adit 600 or * a voip gateway that multiplexes many fxs ports into one ethernet connection. But before I would go down that route I would have to be absolutely sure that the SIP conforms to the standard, the upgrades are free and the fxs ports are compatible with uk standard two wire phones. I found that some two wire phones actually use 4 wires - confusing * a bank of ATAs (handytone 286 or similar). I *really* don't like that solution, as it is a bad botch job and throws lots of issues like which REN they have, many power supplies (or one big one). I really ought to be red in the face for even mentioning that solution. But if nothing else is available, I would probably have to buy them in bulk, take the boards out and mount them in a 19'' box together with a hub so I build my own voip gateway :) maybe it's not so botch after all :) ) For connection to the PSTN: We have three BT lines, and again, we would not like to move over to a different technology like ISDN. The lines work for us, and 'if it ain't broke, don't fix it'. We would use three Sipura SPA-3000 interfaces to connect them to the internal network. The SPA-3000 is sold in the UK and has the CE approval, so it should legally be ok. I am experimenting with one unit at the moment, and am smacked by the literally hundreds of options it has. But I heard good reports about that one, so I expect it to work well in our setting. Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. I got an Adtran 600 with 12 X FXO and 12 X FXS cards for $495 from Penny Doyen [EMAIL PROTECTED] With the strength of the pound, that would practically be free to you! Chris Mason Date: Fri, 8 Apr 2005 17:42:56 +0200 Maybe following options: 1-) Get another channel bank from ebay at low cost. Which will also need another T1 card; 2-) Use 40 voip phones at 50 USD each and you no longer need the card neither the channel bank. But a reliable local network ; Hello, I am working for a charity in the UK and I am projecting a new phone system. We would like to connect our two-wire telephones (40 or so) to an ADIT 600 channel bank, and connect that into an Asterisk box via the CMG card or T1 card. I have been in talks with Carrier Access about the purchase
RE: [Asterisk-Users] Channel bank replacement
Word of warning, get the version 5 or higher FXS cards with the ADIT600, else you will have echo problems. This is just from personal experience. Supposedly the 5 and higher cards have dynamic impedance adjustment, it's worth it. Matt -Original Message- From: Peter Hoppe [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Channel bank replacement Thank you so much for your answers already, I really appreciate it! I have looked into using an Adtran Total Access 750 platform instead, but got away from that idea after I saw the totally confusing amount of options of different modules I can buy. The Adit 600 seemed so much simpler to put together. Also, the Adit 600 had such an excellent appraisal in the asterisk voip-info - see http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank But maybe I need to come back to the Adtran TA750. Unfortunately that platform seems to only offer 24 fxs ports per unit and I need to buy an expensive T1 card. I would buy the Digium T1 card - it seems that it is by far the least expensive card, but $500 is still something. That's why I toyed with the Adit 600 plus cmg card - all I need is a standard network card on the Asterisk machine. We have sorely abandoned the idea of using an extensive amount of voip phones on the property, as we are not a homogenous office setup (ppl also live on the property). This solution would mean * putting in an entire new cat5 network. I would be the person who would have to put it all in place - When would I be finished? In 2 years? 4 years? 10 years? * lots of admin hassle to enable all the phones / add new phones / remove phones * users can't easily extend stations at end points. With two wire phone they simply switch one parallel to the existing one - no admin hassle / extra hubs etc. * two wire technology enables us to buy almost any phone available. * security concerns with the SIP protocol. See http://secunia.com/advisories/8169/ as an example * users potentially plugging their laptops into the voip sockets and browsing/downloading away = lots of setup/admin hassle with the firewall (how do you block Kazaa?) * Phones potentially breaking when users unplug power during firmware download. For example, this is an issue with the Grandstream phone. The only alternative that seems feasible at the moment would be * a different channel bank than the adit 600 or * a voip gateway that multiplexes many fxs ports into one ethernet connection. But before I would go down that route I would have to be absolutely sure that the SIP conforms to the standard, the upgrades are free and the fxs ports are compatible with uk standard two wire phones. I found that some two wire phones actually use 4 wires - confusing * a bank of ATAs (handytone 286 or similar). I *really* don't like that solution, as it is a bad botch job and throws lots of issues like which REN they have, many power supplies (or one big one). I really ought to be red in the face for even mentioning that solution. But if nothing else is available, I would probably have to buy them in bulk, take the boards out and mount them in a 19'' box together with a hub so I build my own voip gateway :) maybe it's not so botch after all :) ) For connection to the PSTN: We have three BT lines, and again, we would not like to move over to a different technology like ISDN. The lines work for us, and 'if it ain't broke, don't fix it'. We would use three Sipura SPA-3000 interfaces to connect them to the internal network. The SPA-3000 is sold in the UK and has the CE approval, so it should legally be ok. I am experimenting with one unit at the moment, and am smacked by the literally hundreds of options it has. But I heard good reports about that one, so I expect it to work well in our setting. Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. I got an Adtran 600 with 12 X FXO and 12 X FXS cards for $495 from Penny Doyen [EMAIL PROTECTED] With the strength of the pound, that would practically be free to you! Chris Mason Date: Fri, 8 Apr 2005 17:42:56 +0200 Maybe following options: 1-) Get another channel bank
Re: [Asterisk-Users] Channel bank question
Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Channel bank question Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean I am assuming you are in the USA, correct me if incorrect. Correct. You want to call your telco and see what the cost of a PRI (T1) is to replace those 10 lines. You have 10 analog lines should be at the point where it is about a break even, if not call a competitive carrier. Not in my area. I have one provider who is brave enough to ATM a t1 out to my location. Everybody else won't touch us. Currently, we have what our vendor is calling a burstable t1. I don't know if this is a common term or not, but bassically it means voice and data share the t1, voice eating into the bandwidth as needed. The t1 is actually terminated into an Adtran 616 which I am currently researching to see if it can feed out a t1 feed instead of the 10 copper lines. But I digress. You do not want to use a channel bank to convert analog to digitial, even if it could be done you are putting bandaids on a huge wound. Agreed. However, given my options You will get a lot of features with the PRI you can not get on analog, not to mention it will work, what you are talking about doing makes no sense from a practical standpoint. Well, except it's probably the best solution when you consider cost/complexity. Do it right, get a PRI and a single PRI digium card (or another PRI terminating device like a T1 chabbel bank)/ Normally, I'd agree with you. However, this situation is different given the line costs. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank question
You may be in luck!!! The Adtan 600 line does have a DSX-1 module available. (you gotta love Adtran!!) http://www.adtran.com/static/docs/64200612L28.pdf Now all you need are a buch of IP phones and your rocking Trash the CB plan go Digital -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Tuesday, April 05, 2005 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel bank question Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Channel bank question Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean I am assuming you are in the USA, correct me if incorrect. Correct. You want to call your telco and see what the cost of a PRI (T1) is to replace those 10 lines. You have 10 analog lines should be at the point where it is about a break even, if not call a competitive carrier. Not in my area. I have one provider who is brave enough to ATM a t1 out to my location. Everybody else won't touch us. Currently, we have what our vendor is calling a burstable t1. I don't know if this is a common term or not, but bassically it means voice and data share the t1, voice eating into the bandwidth as needed. The t1 is actually terminated into an Adtran 616 which I am currently researching to see if it can feed out a t1 feed instead of the 10 copper lines. But I digress. You do not want to use a channel bank to convert analog to digitial, even if it could be done you are putting bandaids on a huge wound. Agreed. However, given my options You will get a lot of features with the PRI you can not get on analog, not to mention it will work, what you are talking about doing makes no sense from a practical standpoint. Well, except it's probably the best solution when you consider cost/complexity. Do it right, get a PRI and a single PRI digium card (or another PRI terminating device like a T1 chabbel bank)/ Normally, I'd agree with you. However, this situation is different given the line costs. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank question
Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank question
Daylight Saving Time confused me as well!!! I'll make it simple: FXO ports connect to a phone company line, can be referred to as Office FXS ports connect to a phone device, can be referred to as Station What kind of features do you want in a channel bank? Not many!! Good channel banks are pretty simple devices. I prefer and recommend an Adtran 750. It has modules that you can change between FXS and FXO. I would look on EBAY for one with 3 FXO cards and 3 FXS cards, that will give you 12 Office lines and the ability to connect 12 analog devices to it as well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, April 04, 2005 5:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Channel bank question Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank question
On April 4, 2005 06:40 pm, Sean Kennedy wrote: If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? you want an FXO channel bank, or at least a channel bank with 10 FXO channels, since you'll be wiring it up to the telco. While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? You want CPD (calling party disconnect, also know as far end disconnection, disconnect supervision, etc.). On FXS it doesn't matter but on FXO it's a critical feature IMO. Carrier Access ABI and ABII do not have this feature. CAC's Adit600 does. I don't know about the others. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Channel bank question Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean I am assuming you are in the USA, correct me if incorrect. You want to call your telco and see what the cost of a PRI (T1) is to replace those 10 lines. You have 10 analog lines should be at the point where it is about a break even, if not call a competitive carrier. You do not want to use a channel bank to convert analog to digitial, even if it could be done you are putting bandaids on a huge wound. You will get a lot of features with the PRI you can not get on analog, not to mention it will work, what you are talking about doing makes no sense from a practical standpoint. Do it right, get a PRI and a single PRI digium card (or another PRI terminating device like a T1 chabbel bank)/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, April 04, 2005 6:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Channel bank question Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? You need a channel bank that has at least 10 or 12 FXO ports. I recommend an Adtran 750 or 850. You can get them on EBay for around $ 400 to 500. But most are pre-configured with FXS. You will need to either switch some of those card out. Then you just put in a T110p card into the asterisk. Also if you get this C/B with 12 FXO you can have the other 12 with FXS for normal analog extensions. While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank Echo
We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution to this. Thanks. PSTN? -- VOIP Network -- Asterisk -(T1)- channel bank -- analog ^echo heard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank Echo
On Sat, 29 Jan 2005, Matt Schulte wrote: We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution to this. Thanks. PSTN? -- VOIP Network -- Asterisk -(T1)- channel bank -- analog ^echo heard If you hear the echo at the marked analog endpoint then it is almost certainly far end echo. This is nearly almost present when calling an analog phone at the far end. On short links without VoIP the reflected energy will sound like a nice sidetone. For longer links (e.g. international) and VoIP you need an echo canceler in the call path. Since you have an analog phone attached to an endpoint there may be an echo heared from the pstn as well. Both these echos can be reduced by adding an echo canceler that has its tail (i.e. subtracts the right amount of the slightly delayed transmitted signal from the received signal) into both directions. Asterisk can act as such an echo canceler. Asterisk may not be the very best echo canceler available, but it may be good enough. Try it and see. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank Echo
Agh, what I meant was the echo is heard from the PSTN side. It seems echo canceling on the T1 (going to channelbank) does nothing, I'm assuming because the T1 is digital and the channelbank is the traversal from digital to analog. -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: Saturday, January 29, 2005 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel Bank Echo On Sat, 29 Jan 2005, Matt Schulte wrote: We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution to this. Thanks. PSTN? -- VOIP Network -- Asterisk -(T1)- channel bank -- analog ^echo heard If you hear the echo at the marked analog endpoint then it is almost certainly far end echo. This is nearly almost present when calling an analog phone at the far end. On short links without VoIP the reflected energy will sound like a nice sidetone. For longer links (e.g. international) and VoIP you need an echo canceler in the call path. Since you have an analog phone attached to an endpoint there may be an echo heared from the pstn as well. Both these echos can be reduced by adding an echo canceler that has its tail (i.e. subtracts the right amount of the slightly delayed transmitted signal from the received signal) into both directions. Asterisk can act as such an echo canceler. Asterisk may not be the very best echo canceler available, but it may be good enough. Try it and see. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank Echo
On Sat, 29 Jan 2005, Matt Schulte wrote: Agh, what I meant was the echo is heard from the PSTN side. It seems echo canceling on the T1 (going to channelbank) does nothing, I'm assuming because the T1 is digital and the channelbank is the traversal from digital to analog. Still, echo cancelling on the T1 should solve the problem since it is on the digital path from the echo source (the hybrids on the channel bank and on the phone) to the pstn. Perhaps your channel bank is set to the wrong impedance compared to the hybrid in the phone? That can cause an echo more powerful than the cho canceler is designed to handle. You need to set echocancelwhenbridged=yes in the zapata.conf for asterisk to even attempt to cancel echo between two digital links. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank Echo
A T1 is a two way transmission media. There is sound going both ways over a '4 wire' interface. 4-wire means that Xmitt is seperate from Recv. In the channel bank there is a 4 wire to 2 wire conversion. It's this junction that introduces echo as some of the 4 wire recv gets feed back into the 4 wire xmitt direction.Echo cancelling has to take place in the 4 wire media path. Lyle - Original Message - From: Matt Schulte [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 29, 2005 12:55 PM Subject: RE: [Asterisk-Users] Channel Bank Echo Agh, what I meant was the echo is heard from the PSTN side. It seems echo canceling on the T1 (going to channelbank) does nothing, I'm assuming because the T1 is digital and the channelbank is the traversal from digital to analog. -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: Saturday, January 29, 2005 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Channel Bank Echo On Sat, 29 Jan 2005, Matt Schulte wrote: We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution to this. Thanks. PSTN? -- VOIP Network -- Asterisk -(T1)- channel bank -- analog ^echo heard If you hear the echo at the marked analog endpoint then it is almost certainly far end echo. This is nearly almost present when calling an analog phone at the far end. On short links without VoIP the reflected energy will sound like a nice sidetone. For longer links (e.g. international) and VoIP you need an echo canceler in the call path. Since you have an analog phone attached to an endpoint there may be an echo heared from the pstn as well. Both these echos can be reduced by adding an echo canceler that has its tail (i.e. subtracts the right amount of the slightly delayed transmitted signal from the received signal) into both directions. Asterisk can act as such an echo canceler. Asterisk may not be the very best echo canceler available, but it may be good enough. Try it and see. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank Echo
On January 29, 2005 12:31 pm, Matt Schulte wrote: We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution to this. Thanks. PSTN? -- VOIP Network -- Asterisk -(T1)- channel bank -- analog ^echo heard If the echo is being heard on the far side then you are generating it at your hybrid. Your diagram is not clear since it was wrapped. I have used the Adit600 and Access Bank 1s with great success -- they do not generate echo (i.e. nobody we've called has heard echo) -- we do, however, hear echo from time to time, even on our PRI. Asterisk's echo cancellation is either getting disabled accidentally or it is not working worth a damn. :-( You can use Tellabs echo cancellation units on T1 and PRI -- these are carrier-grade hardware echo cancellers. I haven't any experience with them yet (waiting for mine to arrive). -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank for T100P or E100P Digium Cards
Hi everyone, I'm looking for a compatible channel bank to use it with T100P/E100P digium cards and asterisk. Who has good experience with a channel bank compatible with this cards? I would like to know brand and model of it, I'm not looking for a fancy product, just a channel bank that allow me to use it as multiplexer between analog lines and E1/T1 digium cards. Thanks in advance, -- Carlos Clemares Director 58 (0) 212 740-53-12/17 [EMAIL PROTECTED] www.radiumtec.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank for T100P or E100P Digium Cards
Carrier Access Corp ABI's Contact me off-list for pricing information. Garrett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Clemares Sent: Tuesday, October 12, 2004 5:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Channel Bank for T100P or E100P Digium Cards Hi everyone, I'm looking for a compatible channel bank to use it with T100P/E100P digium cards and asterisk. Who has good experience with a channel bank compatible with this cards? I would like to know brand and model of it, I'm not looking for a fancy product, just a channel bank that allow me to use it as multiplexer between analog lines and E1/T1 digium cards. Thanks in advance, -- Carlos Clemares Director 58 (0) 212 740-53-12/17 [EMAIL PROTECTED] www.radiumtec.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank for T100P or E100P Digium Cards
Carlos Clemares wrote: Hi everyone, I'm looking for a compatible channel bank to use it with T100P/E100P digium cards and asterisk. Who has good experience with a channel bank compatible with this cards? I would like to know brand and model of it, I'm not looking for a fancy product, just a channel bank that allow me to use it as multiplexer between analog lines and E1/T1 digium cards. I have used Adtran 750/850 without any problems. But I have only used T1 settings. I have also setup and worked with CAC Channel banks Adit 600 is a good choice as well. The adtran 750 with 24 ports fxs will go around $ 500.00 on ebay. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank for T100P or E100P Digium Cards
On Tuesday 12 October 2004 17:16, Garrett Smith wrote: Carrier Access Corp ABI's You do *not* want ABIs or ABIIs if you're looking for FXO ports. I'm partial to the Carrier Access Adit600 myself. You can get them fairly cheaply off ebay and they're a hardware vendor that doesn't hold contempt for the used-parts market. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank for T100P or E100P Digium Cards
The adtran 750 with 24 ports fxs will go around $ 500.00 on ebay. Recent TA 750 ebay purchases -- not by me :) FXO FXS $$$.$$ --- --- -- 0 12 127.50 0 8 152.50 0 24 202.50 0 24 203.50 0 24 203.50 0 24 203.50 0 24 227.50 0 24 230.00 4 16 237.50 0 24 270.00 0 0 271.50 4 20 280.00 0 24 333.00 0 12 355.00 0 16 399.00 0 20 425.00 If you are patient and don't get emotionally involved, $225 is a good buy. On Tue, 12 Oct 2004, Ariel's Hotmail wrote: Carlos Clemares wrote: Hi everyone, I'm looking for a compatible channel bank to use it with T100P/E100P digium cards and asterisk. Who has good experience with a channel bank compatible with this cards? I would like to know brand and model of it, I'm not looking for a fancy product, just a channel bank that allow me to use it as multiplexer between analog lines and E1/T1 digium cards. I have used Adtran 750/850 without any problems. But I have only used T1 settings. I have also setup and worked with CAC Channel banks Adit 600 is a good choice as well. The adtran 750 with 24 ports fxs will go around $ 500.00 on ebay. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank for T100P or E100P Digium Cards
The trick is to make sure you've got the fxo/fxs cards that you need. John Steve Edwards wrote: The adtran 750 with 24 ports fxs will go around $ 500.00 on ebay. Recent TA 750 ebay purchases -- not by me :) FXOFXS$$$.$$ -------- 012127.50 08152.50 024202.50 024203.50 024203.50 024203.50 024227.50 024230.00 416237.50 024270.00 00271.50 420280.00 024333.00 012355.00 016399.00 020425.00 If you are patient and don't get emotionally involved, $225 is a good buy. On Tue, 12 Oct 2004, Ariel's Hotmail wrote: Carlos Clemares wrote: Hi everyone, I'm looking for a compatible channel bank to use it with T100P/E100P digium cards and asterisk. Who has good experience with a channel bank compatible with this cards? I would like to know brand and model of it, I'm not looking for a fancy product, just a channel bank that allow me to use it as multiplexer between analog lines and E1/T1 digium cards. I have used Adtran 750/850 without any problems. But I have only used T1 settings. I have also setup and worked with CAC Channel banks Adit 600 is a good choice as well. The adtran 750 with 24 ports fxs will go around $ 500.00 on ebay. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank for asterisk
On Wednesday 18 August 2004 07:14, Eran Gal wrote: Does anyone know which channel banks work well with asterisk. I've used the Carrier Access Access Bank I and the Carrier Access Adit600. I *far* prefer the Adit600, even though it has an oddball form factor. (It's about 2U tall but only about 6 or 7 inches wide instead of 19). It can handle two T1s and is modular so you can have any combination of FXS and FXO ports (in groups of 8). The Access Bank I and II (II is the 2-T1 version of the ABI) work fine as FXS channel banks, but their FXO modules do *not* detect far-end disconnect which makes them practically useless for terminating FXO. I believe others have used the Adtran TA750 or something along those lines, but I'll let them comment, as I've never seen one. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank
You can use VoiceTronix boards. Joe Pukepail wrote: Since it doesn't look like any of the FXS cards supported by asterisk support analog DID trunks, would it work if I used a T100P connected to an adtran channel bank (atlas 550?) with an FXS card installed? Anyone ever try this configuration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank
Yes you can do it, I've done it with a T100P and an Adtran 612, if you need specific help let me know, look up adtran on the wiki for a similar example. Mitchel On Fri, 13 Aug 2004 20:16:20 -0300, Daniel Bichara [EMAIL PROTECTED] wrote: You can use VoiceTronix boards. Joe Pukepail wrote: Since it doesn't look like any of the FXS cards supported by asterisk support analog DID trunks, would it work if I used a T100P connected to an adtran channel bank (atlas 550?) with an FXS card installed? Anyone ever try this configuration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank
Since it doesn't look like any of the FXS cards supported by asterisk support analog DID trunks, would it work if I used a T100P connected to an adtran channel bank (atlas 550?) with an FXS card installed? Anyone ever try this configuration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank or IAD with message light capability?
I have a possible asterisk application in a hotel/motel situation, where they have several analog phones with message-waiting lights running on an old Mitel PBX. These are neon lights that are illuminated by increasing the on-hook voltage from a nominal 48 to 90 volts DC. Is there on the market either an internet access device or channel bank that can accommodate these phones and control the message waiting indicator from asterisk? -- Jay Hennigan - CCIE #7880 - Network Administration - [EMAIL PROTECTED] WestNet: Connecting you to the planet. 805 884-6323 WB6RDV NetLojix Communications, Inc. - http://www.netlojix.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank Newbie Problem
Hi all I'm trying to configure a TE410P in Europe with three E1s and a T1 channel bank (already bought in the US) to receive and make calls through *. I managed to get the E1s to work, but I'm having trouble with the channel bank (a Rhino). I have tried the bank in spans 1 and 4 changing jumpers in the card (always with no PRI connected, as I have no PRI where I'm testing now, so I'm trying to make the bank work alone). I've tried AutoT1 and also configuring it myself, as ESF and D4, with B8ZS and AMI, changing ZAPTEL and ZAPATA. When I plug the channel bank, it goes ok, framing and signalling lighting green, but after a few seconds, signalling starts blinking, alarm lights red, with display saying Loss of carrier - (LOS - no T1), Asterisk and zttool detecting yellow alarm. The upper right corner of the displays says WINK ESF. Any ideas? Thanks in advance! ZAPTEL.CONF: span=1,1,0,ccs,hdb3,yellow span=2,2,0,ccs,hdb3,yellow #2,2,0 span=3,3,0,ccs,hdb3,yellow #3,3,0 span=4,0,0,esf,b8zs bchan=1-15,17-31 bchan=32-46,48-62 bchan=63-77,79-93 dchan=16,47,78 fxoks=94-117 loadzone=us #es defaultzone=us #es ZAPATA.CONF [channels] switchtype=euroisdn ;pridialplan=unknown signalling=pri_cpe context=default group=1 channel = 1-15,17-31 channel = 32-46,48-62 channel = 63-77,79-93 signalling=fxo_ks context=Internas group=2 channel=94-117 I have also tried things like: ;language=es;context=default;switchtype=euroisdn;rxwink=300;usecallerid= yes;hidecallerid=no;callwaiting=yes;usecallingpres=yes;callwaitingcaller id=yes;threewaycalling=yes;transfer=yes;cancallforward=yes;callreturn=ye s;echocancel=no;echocancelwhenbridged=yes;rxgain=0.0;txgain=0.0 ZTTOOL says: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) ... Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) ... Channel 92: Individual Clear channel (Default) (Slaves: 92) Channel 93: Individual Clear channel (Default) (Slaves: 93) Channel 94: FXO Kewlstart (Default) (Slaves: 94) Channel 95: FXO Kewlstart (Default) (Slaves: 95) ... Channel 116: FXO Kewlstart (Default) (Slaves: 116) Channel 117: FXO Kewlstart (Default) (Slaves: 117) 117 channels configured. TE410P: Span 1 configures for CCS/HDB3 SPAN 1: Primary Sync Source TE410P: Span 2 configures for CCS/HDB3 TE410P: Span 3 configures for CCS/HDB3 TE410P: Span 4 configures for ESF/B8ZS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank problem via long cable
On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote: On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote: Try this if possible. Connect the channel bank to * via the 400' cable, but in the same room as the * box, with all the cable coiled on the floor. Next best thing: I took a coil of 200' of cable and used it to connect the working channel bank that's 200' away to its patch bay. Lo and behold, same problem. I also patched this rig into the port that's supposed to be configured with an LBO of 399'+ and I still get the same problem. I believe this rules out location and leaves the length of the cable as the culprit. This is all using cat6 cable, fwiw. I'm really beginning to suspect the TE405P isn't cranking up the LBO like it's supposed to. I've asked Digium for some insight and hope to hear back from them today. Anyone have any recommendations for good T1 extenders? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank problem via long cable
On Thu, Jun 24, 2004 at 09:34:02AM -0400, Timothy R. McKee wrote: Looking back, I see you are running B8ZS/ESF. I ran into similar problems with a 100' run to a CAC AB-II. As soon as I switched to AMI/D4(SF) all my problems went away. Did I say that? I've actually been running AMI/D4 to my other channel banks, and right now I'm running AMI/D4 the channel bank that's giving me trouble as well. I tried B8ZS/ESF a few times, but that didn't help so I switched it back. (:.:) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank problem via long cable
Probably an impedance problem. PCM line signals are designed to be transmitted over a *twisted telephone cable* having 120 ohms at 1 MHz. I'm not sure that cat6 cable fulfil this requirement. Maybe cat3. Jorge Bonzo Armstrong wrote: On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote: On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote: Try this if possible. Connect the channel bank to * via the 400' cable, but in the same room as the * box, with all the cable coiled on the floor. Next best thing: I took a coil of 200' of cable and used it to connect the working channel bank that's 200' away to its patch bay. Lo and behold, same problem. I also patched this rig into the port that's supposed to be configured with an LBO of 399'+ and I still get the same problem. I believe this rules out location and leaves the length of the cable as the culprit. This is all using cat6 cable, fwiw. I'm really beginning to suspect the TE405P isn't cranking up the LBO like it's supposed to. I've asked Digium for some insight and hope to hear back from them today. Anyone have any recommendations for good T1 extenders? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank problem via long cable
Bonzo Armstrong wrote: On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote: On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote: Try this if possible. Connect the channel bank to * via the 400' cable, but in the same room as the * box, with all the cable coiled on the floor. Next best thing: I took a coil of 200' of cable and used it to connect the working channel bank that's 200' away to its patch bay. Lo and behold, same problem. I also patched this rig into the port that's supposed to be configured with an LBO of 399'+ and I still get the same problem. I believe this rules out location and leaves the length of the cable as the culprit. This is all using cat6 cable, fwiw. I'm really beginning to suspect the TE405P isn't cranking up the LBO like it's supposed to. I've asked Digium for some insight and hope to hear back from them today. Looks like you're doing the right thing in regards to troubleshooting, only a line analyzer will tell you the whole story. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank problem via long cable
On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote: Nah, a true analyzer will detect framing errors do loopback echo tests, etc. After much pfutzing around and talking with CAC's tech support, I'm finally coming around to your original suggestion and am in the process of finding someone to rent/loan me an analyzer or come out and sniff the line for me. My current suspicion (which I'll need an analyzer to confirm or refute) is that the LBO parameter in zaptel.conf is not actually having an effect. I have to think that if this was the case for this card in general, someone else would have run into it before now and would have mentioned it, but I don't see any references to any such problems in the archives. But I am curious whether there are people out there successfully driving their TE405P over more than 400' of cable. Try this if possible. Connect the channel bank to * via the 400' cable, but in the same room as the * box, with all the cable coiled on the floor. I haven't tried this yet as I don't have a 400' cable handy. But it just now occurred to me that I have a box at home that might have 400' left in it so I may be able to try it tomorrow. I can confirm that I can talk to at least two of my three access banks with no problem at all if they're plugged in at 200'. ZTTool also has a loopback test if I'm not mistaken. It may give you some insight. Looping the line up doesn't tell me much. Neither does plugging a loopback plug in at either end of the line. No matter how I loop it, neither end shows alarms. Very odd. Hopefully I'll be able to find an analyzer tomorrow. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank problem via long cable
I've got a Digium TE405P feeding three Carrier Access AB-II units located in three separate suites of the complex my company resides in. I've had one AB-II connected over about 15' of cable and running for several weeks now, and it seems to be perfectly happy. Over the weekend, I added two more AB-IIs, one in each of the other two suites. The second unit is about 150 cable feet away from the switch and it syncs up just fine, but the third is 400 cable feet away and is having problems with its T1. The dip switches on all three units are set the same, and with the exception of the LBO they're all configured the same in zaptel.conf, yet the furthest AB-II is showing errors. I've swapped AB-IIs between positions to no effect; whichever unit is in the furthest suite cannot fully sync. There are at least two cables available in each segment of the run between the far AB-II and the switch (due to the way our complex is wired, the run is a total of three in-wall cables and four patch cables), and I have tried them all. I have also tried plugging into other ports on the TE405P. I've tried changing the signalling and the framing. I've tried changing the LBO value to everything from 0 to 4. Through all this, the constant appears to be that any AB-II placed 400' away from the switch will not behave. When initially connected, the led next to the T1 jack on the AB-II blinks yellow a few times, then turns green for at most a quarter second and then to solid yellow where it stays. The other status led on the AB-II for the span stays a solid green. Picking up a phone connected to this unit gets me a very noisy dialtone. If I dial a single digit to get rid of the dialtone, I hear a pretty steady buzzing noise where there should be silence. The curious thing here is that if I use this phone to dial another extension, 1) there is no problem dialing and 2) if I use a station that's working properly to listen to the voicemail message, I hear a perfectly clear signal with just a hint of that same buzzing in the background. My zaptel.conf looks like this: span=1,1,0,d4,ami span=2,1,0,d4,ami span=3,1,4,d4,ami fxoks=1-16 fxoks=17-24 fxoks=25-48 fxoks=49-72 Can anyone tell me what I'm missing or what else I might try? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank problem via long cable
Do you have access to a T-1 analyzer? You more than likely have a 'dirty' T-1 line that is out of spec based on the length of the run. Nik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bonzo Armstrong Sent: Monday, June 21, 2004 5:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Channel bank problem via long cable I've got a Digium TE405P feeding three Carrier Access AB-II units located in three separate suites of the complex my company resides in. I've had one AB-II connected over about 15' of cable and running for several weeks now, and it seems to be perfectly happy. Over the weekend, I added two more AB-IIs, one in each of the other two suites. The second unit is about 150 cable feet away from the switch and it syncs up just fine, but the third is 400 cable feet away and is having problems with its T1. The dip switches on all three units are set the same, and with the exception of the LBO they're all configured the same in zaptel.conf, yet the furthest AB-II is showing errors. I've swapped AB-IIs between positions to no effect; whichever unit is in the furthest suite cannot fully sync. There are at least two cables available in each segment of the run between the far AB-II and the switch (due to the way our complex is wired, the run is a total of three in-wall cables and four patch cables), and I have tried them all. I have also tried plugging into other ports on the TE405P. I've tried changing the signalling and the framing. I've tried changing the LBO value to everything from 0 to 4. Through all this, the constant appears to be that any AB-II placed 400' away from the switch will not behave. When initially connected, the led next to the T1 jack on the AB-II blinks yellow a few times, then turns green for at most a quarter second and then to solid yellow where it stays. The other status led on the AB-II for the span stays a solid green. Picking up a phone connected to this unit gets me a very noisy dialtone. If I dial a single digit to get rid of the dialtone, I hear a pretty steady buzzing noise where there should be silence. The curious thing here is that if I use this phone to dial another extension, 1) there is no problem dialing and 2) if I use a station that's working properly to listen to the voicemail message, I hear a perfectly clear signal with just a hint of that same buzzing in the background. My zaptel.conf looks like this: span=1,1,0,d4,ami span=2,1,0,d4,ami span=3,1,4,d4,ami fxoks=1-16 fxoks=17-24 fxoks=25-48 fxoks=49-72 Can anyone tell me what I'm missing or what else I might try? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank problem via long cable
On Mon, Jun 21, 2004 at 12:17:38PM -0500, Nik Martin wrote: Do you have access to a T-1 analyzer? You more than likely have a 'dirty' T-1 line that is out of spec based on the length of the run. Sadly, none that I'm aware of, but I'll ask around. I could probably find a decent scope to put on the line, but I'm not sure what I'd be looking for. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel bank problem via long cable
On Mon, Jun 21, 2004 at 12:17:38PM -0500, Nik Martin wrote: Do you have access to a T-1 analyzer? You more than likely have a 'dirty' T-1 line that is out of spec based on the length of the run. Sadly, none that I'm aware of, but I'll ask around. I could probably find a decent scope to put on the line, but I'm not sure what I'd be looking for. Nah, a true analyzer will detect framing errors do loopback echo tests, etc. Try this if possible. Connect the channel bank to * via the 400' cable, but in the same room as the * box, with all the cable coiled on the floor. Does it work? If yes, than you need better cable shielding, etc. on the cable run. If not, replace the 400' with say, 300' and see if it works. ZTTool also has a loopback test if I'm not mistaken. It may give you some insight. Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank Frustrations
I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime green. The T100P is sometimes red, sometimes green. The zaptel configuration is: span=1,0,0,esf,b8zs The channel bank settings are: Clock source: On or Off, seems not to make a difference Framing: ESF Line Code: B8ZS CSU: On or Off, seems not to make a difference Once I managed to get all three status lights on the channel bank green but the T100P was red. On the few cases the T100P gives a green status, zttool shows the RxB bits randomly flipping one and off. I have tried different T100P cards in different servers so that has been eliminated as a cause. I have made up several T1 loopback cables so I don't think it is a flaky cable. The remaining possibilities are: a) a bad channel bank (although it passes the self test) b) a bad zaptel configuration After several hours of trying different settings and DIP switches I am increasingly frustrated in trying to determine the cause of the problem. It has been especially difficult since power cycling the channel bank can result in a change to the status lights without any change to the settings or configuration. Any suggestions on what might be causing the problem or what to try next? We're trying to go live this week and this problem is critical. Thanks for any help/suggestions g. P.S. - What, exactly, is the meaning of the second argument to the zaptel span parameter (timing source) and how does it relate to the clock switch on the channel bank as I have tried all four combinations without obvious consistent affect? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank Frustrations
George, We have this config working. Please give me a call (yeah, I'm at the office too) and we can walk through your config together. -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: George Pajari [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 5:51 PM Subject: [Asterisk-Users] Channel Bank Frustrations I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime green. The T100P is sometimes red, sometimes green. The zaptel configuration is: span=1,0,0,esf,b8zs The channel bank settings are: Clock source: On or Off, seems not to make a difference Framing: ESF Line Code: B8ZS CSU: On or Off, seems not to make a difference Once I managed to get all three status lights on the channel bank green but the T100P was red. On the few cases the T100P gives a green status, zttool shows the RxB bits randomly flipping one and off. I have tried different T100P cards in different servers so that has been eliminated as a cause. I have made up several T1 loopback cables so I don't think it is a flaky cable. The remaining possibilities are: a) a bad channel bank (although it passes the self test) b) a bad zaptel configuration After several hours of trying different settings and DIP switches I am increasingly frustrated in trying to determine the cause of the problem. It has been especially difficult since power cycling the channel bank can result in a change to the status lights without any change to the settings or configuration. Any suggestions on what might be causing the problem or what to try next? We're trying to go live this week and this problem is critical. Thanks for any help/suggestions g. P.S. - What, exactly, is the meaning of the second argument to the zaptel span parameter (timing source) and how does it relate to the clock switch on the channel bank as I have tried all four combinations without obvious consistent affect? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank - Vina T-1 Integrator
Has anyone tried a Vina T-1 Integrator as a channel bank with Asterisk? They appear to be plentiful, but I want to make sure I'm not buying a brick. THX/BDH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank - New * install
On Thu, 22 Apr 2004, Steven Critchfield wrote: ... VoIP phones have the benefit of linear growth cost. A phone costs $X, and for the most part will cost $X no matter how many lines you roll out. So a new extension is just $X increase, and your system is just $X x N extensions to deploy. Also VoIP can be deployed pretty much anywhere. Analog has the benefit of cheaper phones, and what I consider a better service record. There isn't really a problem of what is and isn't supported, or supported to what extent. Draw backs are you can either deploy in multiples of 4 with the TDM400 or go T1 and deploy in multiples of 24. Either way, it makes the first step beyond the current block slightly expensive, but then the increment is a small amount till you fully deploy your current block. 24 Budgetones would be ~$1800(assuming you find them for $75 each). 24 analog ATT phones, $1720(assuming a channel bank from ebay at $500 and $30 phones). So you can see where the 25th phone goes back to the VoIP phones as the 25th phone on analog will run you another $1030 for the T1 port and channel bank. There are other factors too. VOIP phones will have useful diplays an additional buttons (transfer, hold, etc.), plus usually have many call appearances. In addition, VOIP phones offload codec load off your * server, and come with an included G.729a codec. This might not be an issue for you if you use TDM in and out, but if you have a VOIP gateway, VOIP phones start to look good. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank - New * install
I am looking at installing * as the PBX in a new office and have a few questions that I hope someone can help me with. The installation will be small at first with about 8 internal extensions, but will grow to 24 within a year or so. First is there any benefit to using VoIP phones instead of installing a channel bank and analog business phones? If not, what are some good analog business phones that people have used? How about channel banks, can I get some suggestions? Thanks - -Jon Brandon VP of Technology Monsoon Add me to your contacts: http://www.monsoonretail.com/vcards/JonBrandon.vcf
Re: [Asterisk-Users] Channel Bank - New * install
On Thu, 2004-04-22 at 12:50, Jon Brandon wrote: I am looking at installing * as the PBX in a new office and have a few questions that I hope someone can help me with. The installation will be small at first with about 8 internal extensions, but will grow to 24 within a year or so. First is there any benefit to using VoIP phones instead of installing a channel bank and analog business phones? If not, what are some good analog business phones that people have used? How about channel banks, can I get some suggestions? Dude, drop the HTML, and remember why google exists. VoIP phones have the benefit of linear growth cost. A phone costs $X, and for the most part will cost $X no matter how many lines you roll out. So a new extension is just $X increase, and your system is just $X x N extensions to deploy. Also VoIP can be deployed pretty much anywhere. Analog has the benefit of cheaper phones, and what I consider a better service record. There isn't really a problem of what is and isn't supported, or supported to what extent. Draw backs are you can either deploy in multiples of 4 with the TDM400 or go T1 and deploy in multiples of 24. Either way, it makes the first step beyond the current block slightly expensive, but then the increment is a small amount till you fully deploy your current block. 24 Budgetones would be ~$1800(assuming you find them for $75 each). 24 analog ATT phones, $1720(assuming a channel bank from ebay at $500 and $30 phones). So you can see where the 25th phone goes back to the VoIP phones as the 25th phone on analog will run you another $1030 for the T1 port and channel bank. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank?
Four or five analog lines can be done with a single computer so no channel bank is needed. If you need 6 or more than there is also the choice of using two machines and IAX. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime Lanning Sent: Tuesday, April 06, 2004 12:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Bank? quote who=Ken Hello, I'm new to Asterisk and would like to know how you could have 4 to 6 incoming analog POTS lines connecting to the Asterisk server and have 4 to 6 analog lines going out.(A T1 line is too costly). Would 2 channel banks be used? A T1 channelbank has 24 channels, so only 1 is needed. FXO channels (What you connect to the POTS lines) can be both inbound and outbound. If you are not using DID. So, you just need to find out how many concurrent calls you need to support. If you are using analog DID lines, then those are inbound only, and require FXS ports. (You supply dialtone and battery, the carrier's switch picks up your line and dials into your PBX.) Now, there are multiple ways to get the analog lines into Asterisk... o use an external gateway... POTS - SIP - Asterisk o wait until next month and get the FXO multiport cards from Digium o get a T1 card + channelbank -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank?
quote who=John Vogel Four or five analog lines can be done with a single computer so no channel bank is needed. If you need 6 or more than there is also the choice of using two machines and IAX. Talk about port density issues. So, if he really needs all 12 lines, then he needs 3 PCs? (He probably doesn't need all 12.) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
