Re: [asterisk-users] Idiot questions
Thanks for all the replies. I take it that with one of these FXO boards, one would need an IP phone as there is no FXS ? BTW, cheapest I've seen is $19.95. Still, not bad. joea Nilesh Londhe[EMAIL PROTECTED] Boldly Declared: 8/24/2006 8:47 PM: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, Adam Collard [EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: joea, j4computers [EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adam Collard President Digital Telecom of Michigan, Inc. [EMAIL PROTECTED] (517) 233-1072 Direct Office (800) 420-3803 x4101 Office (517) 766-5902 Fax This email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately. ___ --Bandwidth 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] Idiot questions
So: The FXO car is for the Pots lines (I.E. bellsouth line) so if you need a analog phone cennected to asterisk you need a FXS card, so if you gonna use a SIP Soft Phone (or a regular SIP Phone) you only need a network connectivity between Asterisk and SIP Phone. Cris. From: joea, j4computers [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Idiot questions Date: Thu, 24 Aug 2006 21:04:39 -0400 Thanks for all the replies. I take it that with one of these FXO boards, one would need an IP phone as there is no FXS ? BTW, cheapest I've seen is $19.95. Still, not bad. joea Nilesh Londhe[EMAIL PROTECTED] Boldly Declared: 8/24/2006 8:47 PM: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, Adam Collard [EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: joea, j4computers [EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adam Collard President Digital Telecom of Michigan, Inc. [EMAIL PROTECTED] (517) 233-1072 Direct Office (800) 420-3803 x4101 Office (517) 766-5902 Fax This email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately. ___ --Bandwidth 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 _ Check the weather nationwide with MSN Search: Try it now! http://search.msn.com/results.aspx?q=weatherFORM=WLMTAG ___ --Bandwidth 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] Phone status
Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQLtable. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: StatusPrivilege: CallChannel: SIP/310-08697fb8CallerID: 310CallerIDName: unknownAccount: State: UpLink: SIP/311-0868fd98Uniqueid: 1156442804.74 Event: StatusPrivilege: CallChannel: SIP/311-0868fd98CallerID: 311CallerIDName: SnomAccount: State: UpContext: macro-vmExtension: sPriority: 5Seconds: 13Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: StatusPrivilege: CallChannel: SIP/311-08695698CallerID: 35254390CallerIDName: unknownAccount: State: UpLink: IAX2/MR-1Uniqueid: 1156442974.76 Event: StatusPrivilege: CallChannel: IAX2/MR-1CallerID: 35436121CallerIDName: unknownAccount: State: UpContext: macro-vmExtension: sPriority: 5Seconds: 9Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: [asterisk-dev] Phone status
Umm Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQLtable. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --Bandwidth 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] Trunk with multiple IPs?
Benjamin Lawetz wrote: Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? Here's an iax.conf example of what I'm using: [teliax] context=teliax-incoming type=user auth=md5 secret=mysecret jitterbuffer=yes disallow=all allow=gsm deny=0.0.0.0/0.0.0.0 permit=207.174.202.0/255.255.255.0 The last two statements essentially restrict incoming calls from teliax to one of their class-c networks (regardless of how many servers or IP's they have). Note that on incoming calls the host= line is not used. If you're really asking how to do that for outgoing calls, you'll probably have to do it through three/four sections (type=peer) and deal with those sections in your dialplan. As a side note, there are a large percentage of * implementors that don't understand the search terms when an incoming call is being negotiated (eg, is host= used, is secret= used). Without that understanding, calls likely come into different sections then what the implementor actually expected. The deny permit statements are very useful to tighten down security for each incoming context. ___ --Bandwidth 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] Phone status
Hello, I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) snip I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQL table. snip Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? It is far less stable if you perform queries every second than just pasively listening to the events. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth 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] Basic Asterisk Setup
On Jul 5, 2006, at 7:58 AM, K Y Iyer wrote: Hi Hi! Am a bit confused about the basic requirements for a simple, small, test Asterisk setup. There are many options... I want to setup a PBX with 8 PSTN lines and 50 extensions. For argument's sake we'll assume all 50 extensions and 8 PSTN lines could be active at some point in the day. What kind of hardware requires to be inside the Asterisk system for this? This is a vague and unclear question. You will need enough CPU power to handle whatever system design you choose. There are internal cards (PCI) that could hook up the PSTN lines for you, or you could use external hardware (gateway) for that. I understand that the client end will be IP - so no hardware required for this. OK. I can purchase Sangoma in India - I am sure - maybe even Dialogic. I am totally confused about FXSs and FXOs. FXS's hook to phone hand sets (old school type). FXOs hook to PSTN lines. Your above description needs 8 FXO's and NO FXSs, presuming you use IP phones as extensions. Hope this helps. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones locking up
We're running several 320s on 6.2.2 and haven't had any issues just to toss my two cents in. On Thu, 24 Aug 2006 23:19:48 +0200 (SAST), garth wrote Hi All I have had a problem with a few Snom 320's on several sites locking up after a few days. I am running application ver 6.2.2 with the latest jffs2 ver and tried the latest 5.x ver with similar results. Is this also experienced with other Snom users? I know some posts say it could be the network switches etc, but Cisco? I fail to see how a switch could bring down a device. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Attempt to setup paging and intercom
I do not know if this breaks anything or not the way you have it, but you
should not have the underscore before the extension.
The underscore means that the following is an expression, where X=any single
digit and .=any number of digits.
I do not know if the underscore also interprets the * as something, or maybe it
just gets stuck trying to figure out an expression
with no X nor .
Or this may not be an issue at all.
--
--
Steven
http://www.glimasoutheast.org
Larry Alkoff [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
This is my first attempt to setup intercom and paging for some Grandview sip
phones per instructions from Grandview.
I put the lines below in extensions.conf and did the CLI reload command.
When I issue
**1 or **2 from a phone I get a 404 error.
Shouldn't that be ringing the 3 phones on my list?
The instructions are a little vague (to a newbie like me) and may well be
wrong.
Here is what I put in extensions.conf:
-- Stop reading here if not interested
; from: FAQ_Asterisk_Paging_for_GXP-2000.pdf
; Paging and Intercom:
;
; Grandstream Phone Configuration:
; Allow Auto Answer by Call-Info: Yes
; Turn off speaker on remote disconnect: Yes
; Note: Above configuration will allow GXP-2000 to auto answer a call
; when the call contains:
; SIP header Call-Info: answer-after=0
; And when the call hung up by the remote party,
; the phone will automatically on hook without alerting user with
; disconnect busy tones.
; Asterisk Configuration:
; ===
; Then you can set up Asterisk with following functions:
; 1) One to One Intercom
; ==
; You will first define a Macro and then use it in the one to one intercom
context:
[macro-pageext]
exten = s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for ANY call
exten = s,2,SIPAddHeader(Call-Info: answer-after=0)
exten = s,3,Dial(${ARG1})
exten = s,4,NoOp() ; Add others here
exten = s,5, Hangup
exten = s,102,Hangup
[INTERCOM_GROUP]
exten = _*1XX,1,Macro(pageext,SIP/${EXTEN:1}) ;Page each extension
exten = _*1XX,2,Hangup
; Note: Above configuration will allow user intercom with any extension
; (using 1XX) by dialing *1XX.
; 2) One to Many Paging
; =
[One_Way_Page_GROUP]
exten = _**1,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**1,2,Page(${One_Way_Paging_List}|)
exten = _**1,3, Hangup
; Note: Above configuration will allow user to one way page(broadcast)
; to all
; the extensions defined in variable One_Way_Paging_list
; which can be define as following:
One_Way_Paging_List = SIP/120SIP/122/SIP/100
; 3) One to Many Intercom
; ===
[Two_Way_Intercom_GROUP]
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(${Two_Way_Intercom_List}|d)
exten = _**2,3, Hangup
; Note: Above configuration will allow user to do two way intercom to all the
; extensions defined in variable Two_Way_Intercom_List which can be
; define as following:
Two_Way_Intercom_List = SIP/120SIP/122/SIP/100
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] Can the codec/format for name/greeting in voicemail be changed?
Hello people,
before I go hunting on Wiki and Google, if maybe someone here knows
the answer to this.
This is in regards to the voicemail system. Is it possible to change
the default/native format in which the greetings and outgoing messgaes
for a user's mailbox are stored? It seems like (*) records everything
in a GSM 6.10, mono 8kHz. If I was using the filesystem, then I could
run a cronjob or something and convert all greetings etc. in the
formats that I expect the endpoints to be using, but since I use
Realtime, I don't have that luxury.
How then can I get (*) to either record in a different format OR be
able to convert these voicemail subscriber greetings in my database to
some other format?
Any ideas? suggestions?
Thanks in advance,
${RR}
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[asterisk-users] Re: Intercom mode on Polycom and/or SPA9xx
Thanks to BJ Weschke I have now solved this problem by adding the option s, and taking off the option t from app_page like this: I changed the line that reads (by me line 177): snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid, ast_test_flag(flags, PAGE_DUPLEX) ? : m); to: snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdsw, confid, ast_test_flag(flags, PAGE_DUPLEX) ? : m); and the line that reads (by me line 192): snprintf(meetmeopts, sizeof(meetmeopts), %ud|A%sqxd, confid, ast_test_flag(flags, PAGE_DUPLEX) ? : t) to: snprintf(meetmeopts, sizeof(meetmeopts), %ud|A%sqxd, confid, ast_test_flag(flags, PAGE_DUPLEX) ? : ); Now the one being paged can dial *1 to unmute them selfs, and then the caller (in our case the person paging) can complete the transfer, and it all works. Thank you BJ On 7/11/06, C F [EMAIL PROTECTED] wrote: I have a customer that is used to Intercom from ther Avaya system, where you just page someone and until they don't pick up the handset there is only one way audio from the caller to callee. At the moment I'm using for the Polycom ALERTINFO to a customized ring that auto answers, and for the Sipura spa941 SipAddHeaders that also autoanswers however they both do 2 way audio, is there anyway that it can be configured to 1 way audio? I know that I can do meetme with mute, but that wont work for 3 reasons: 1. Unmute will only work with a DTMF, which I realy want handset to do it. 2. Xfers wont work as I want, since I'm using the intercom on attxfers. 3. My boxes might or might not have the power to handle many meetmes, and I don't want to run into this. Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status not updating on inbound
Hints have got a lot better over the last year, so upgrading might be the only was to get this working really well. Kind regards, PaulH AsteriskIT www.asteriskit.com.au On Thu, 2006-08-24 at 17:36 -0600, Damon Estep wrote: I have the “hint” priority defined for a few SIP phones. When I make a call OUT from one of the phones I see that the “show hints” picks up a status change from 0 to 1 for the extension, but when I call IN to that extension the hint status is still 0. This is on a server built back in September of 2005. It has been very reliable (and busy) so I do not want to upgrade it if I can avoid it. Does anyone know if this was a bug at one time? Could there be something in my config that is causing this behavior? ___ --Bandwidth 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
