Re: [asterisk-users] wireless VOIP phone recommendations?
A linksys PAP2 with a Motorola Dect set is what I use for a wireless IP phone solution. I have tried Zyxel y Linksys wifi phones, and a couple of others, but the battery life just isn't workable on WIFI phones. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de marvin horst Enviado el: martes, 05 de febrero de 2008 14:53 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] wireless VOIP phone recommendations? I have been using the D-Link DPH-540 wireless VOIP handset, and I really like this phone. We had tried the UStarcomm phone, but the phone is used in a noisy environment and the volume wasn't loud enough. The problem with the D-Link phone is the Li-ion battery needs to be replaced and D-Link doesn't sell a replacement battery and I haven't found any after-market batteries. So this phone is essentially a brick because I need a new battery :( So any recommendations for another wireless VOIP phone? -- Marvin Horst ___ -- 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] Upgrade to Asterisk 1.4 - it's one year's old!
I think it should be core dogs show black. Seriously though, I think you make a good point. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Steve Edwards Enviado el: miercoles, 19 de diciembre de 2007 4:43 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! On Sat, 15 Dec 2007, Johansson Olle E wrote: I wonder if there are any major obstacles for upgrading. How about the change from a bad command line interface to a really bad command line interface? I mean, Seriously? (in a Grey's Anatomy kind of way...) The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. The new syntax continues down the noun-verb path instead of correcting itself and using verb-noun like most other applications (MySQL, GDB, Oracle, etc.) Then, just to make it worse, now I have to learn which commands somebody (arbitrarily) decided are core and which are not -- for what benefit? Certainly doesn't make MY job easier! Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Is it too late to fix this for 1.6? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] really sorry about this - E1 vs T1
The cards ship configured for T1. If you didn?t change the jumpers, it is set for T1. If it is set for T1 and you really want an E1 and you configure your zapata.conf as you would for an E1, you will get an error around channel 25, which tells you that you forgot to change the jumpers, and you have to call the guy on site and ask him (again) to close the jumpers on the card This has never happened to me of course, but it happens regularly to this guy that I know.. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Julian Lyndon-Smith Enviado el: viernes, 12 de octubre de 2007 0:08 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] really sorry about this - E1 vs T1 I am *really* sorry about hijacking this thread, but the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, Digium, sort the filters out!) I installed my super-duper new TE412P card today, without remembering to check the settings for T1/E1. As the server is now a hundred miles away, is there a) Any way of checking what setting is in place b) Changing that setting without having to physically remove the card and see ? Julian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Message Logs
You can configure logger.conf so that it will log just about everything you could want. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+logger.conf -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de [EMAIL PROTECTED] Enviado el: jueves, 23 de agosto de 2007 17:15 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Asterisk Message Logs Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through an Asterisk server. I can't seem to get communication between the two phones. Does anyone have any experience using these open-source Jain-sip-applet-phones? Thanks, Denis ___ --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 attachment: winmail.dat___ --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] Multiple servers using realtime
I use a centralized database (with replication) for several servers, and it works very well. I keep all the mysql traffic on a separate network from the SIP traffic. It makes it easy to add capacity. If you are doing all the mySQL on one box anyway, I don?t see any adavantage to using separate databases. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Peder @ NetworkOblivion Enviado el: miercoles, 22 de agosto de 2007 19:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Multiple servers using realtime I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --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] Realtime Queue Members
I have it working fine in 1.4.x, but I also have the queues defined in the Realtime database and not in the queues.conf -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Anthony Francis Enviado el: martes, 21 de agosto de 2007 1:46 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Realtime Queue Members Peder @ NetworkOblivion wrote: Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --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 There is no queue_members file, asterisk doesnt know hat you are talking about, you would have to #include queue_members from inside that queue definition. ___ --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] Dial plan suggestions
Call Park / Call Pickup would probably be the best option for this. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Russell Handorf Enviado el: martes, 14 de agosto de 2007 19:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Dial plan suggestions Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). What I was thinking on how to do this is using some sort of call parking for the hunt group of all the phones in the house. Once the call is picked up, it then places both the SIP phone and caller into a meetme conference room. To simply join that static assigned room, one of the other phones picks up and joins that room. What I have a concern about is if we hang up, the caller can still sit there and listen in. When no phones are active, it should disconnect the caller. Does this implementation make sense? Has anyone else done something like this? Thanks. ___ --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] Dialplan loop
I do something like this. But I am using Realtime and 1.4
context185 1 Set caller_num=${CALLERID(num)}
context185 2 SetMusicOnHold default
context185 3 Playbacksilence2
context185 4 AGI context1.php
context185 5 Set __FWCOUNT=0${FWCOUNT}
context185 6 Set __FWCOUNT=$[${FWCOUNT}+1]
context185 7 GotoIf $[${FWCOUNT}10]?10
context185 8 DialSIP/ext1SIP/ext2|30|m
context185 9 Hangup
context185 10 Congestion
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Anselm
Martin Hoffmeister
Enviado el: domingo, 12 de agosto de 2007 12:35
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Dialplan loop
Am Donnerstag, den 09.08.2007, 20:12 -0500 schrieb David Bandel:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
exten = s,n,Set(loop = 0)
...
exten = s,n,Set(loop = $[${loop} + 1])
The above loop increment doesn't work. The error message is:
WARNING[14490]: ast_expr2.fl:398 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '+', expecting $end; Input:
+ 1
^
WARNING[14490]: ast_expr2.fl:402 ast_yyerror: If you have questions,
please refer to doc/channelvariables.txt in the asterisk source.
Try removing extra space characters around the =. Very similar example
from my dialplan
exten = _2XX,n,Set(I=1)
...
exten = _2XX,n,Set(EXTR=$[${I} + 1])
Works fine. Also assigning a variable a new value based on the old value
works OK here (although not calculated, but concatenated):
exten = _2XX,n,Set(D=${D}SIP/sip501)
I am using Asterisk 1.2 here, but I remember similar errors with stray
characters.
BR
Anselm
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[asterisk-users] test list
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Re: [asterisk-users] How strip +1 from caller id on inbound call
What if it is an international call? Then your callerID won't work.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: lunes, 13 de agosto de 2007 3:21
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How strip +1 from caller id on inbound call
After rereading this post, I belive that this could also be
acomplished doing this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than 10 digits grab the last 10 digits of the CIDNUM
exten = _X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):-10})
;this grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
On 8/12/07, C F [EMAIL PROTECTED] wrote:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-
10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Hope this helps.
On 8/12/07, voiplist [EMAIL PROTECTED] wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME}
${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
--
Prestige Messaging
Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com
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[asterisk-users] Shared Line Appearance - Aastra 55i - Does it work?
Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4?
Specifically with the Aastra 55i.
Specifically, I am using the Aastra 55i with the expansion module.
We want to see if other handsets are being used. (BLF) Getting BLF to work
would be a great start. It sounds like setting up the hints properly will
achieve this. right? Not totally sure how this should be configured.
We also want bridged appearances. Shared Line Appearance in Asterisk 1.4.
It is my understanding that with a bridged appearance, the line would show
as busy if it is in use on another handset, right? Meaning that the BLF
would be irrelevant?
in SLA.conf we have:
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
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[asterisk-users] Shared Line Appearance - Aastra 55i - Does it work?
Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4?
Specifically with the Aastra 55i.
Specifically, I am using the Aastra 55i with the expansion module.
We want to see if other handsets are being used. (BLF) Getting BLF to work
would be a great start. It sounds like setting up the hints properly will
achieve this. right? Not totally sure how this should be configured.
We also want bridged appearances. Shared Line Appearance in Asterisk 1.4.
It is my understanding that with a bridged appearance, the line would show
as busy if it is in use on another handset, right? Meaning that the BLF
would be irrelevant?
in SLA.conf we have:
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
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Re: [asterisk-users] Learn some terminalogy before mountingthistask.
Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices,
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices,
