Re: [asterisk-users] Advice on GUI
On Sun, 2006-10-29 at 15:33 -0800, Tom Lynn wrote: Without providing a link to the list, or citing your front-runners, you can't really expect people to reply, can you? http://www.voip-info.org/wiki-Asterisk+GUI I am currently trying out VoiceOne. On 10/27/06, Frédéric Blaise [EMAIL PROTECTED] wrote: Hello all I would like to know your opinions on free GUI used to manage Asterisk. Which is better? My setup is quite small, about 15-20 phones. I've seen the liste on voip-info. Thanks all. fred ___ --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 signature.asc Description: This is a digitally signed message part ___ --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] Incorrect Ring tone. Getting a US tone when it should be AU tone
I don't think it is a phone problem. I get a US ring tone on a PAP2, SPA-942 and IDEFdisk softphone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Monday, 30 October 2006 5:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incorrect Ring tone. Getting a US tone whenit should be AU tone What phones are you using? It could be a phone level issue. (my aastra has a setting for AU sounds..) PaulH On Mon, 2006-10-30 at 16:13 +1100, Klaverstyn, David C wrote: For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general] country=au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Information on Asterisk 1.4-beta 3 and ARA
Hi everyone, I'm working with Asterisk 1.4-beta3 and ARA for my Univesity thesis, to enable jingle support into an administrative framework for asterisk developed in our lab. It's possible to map jabber's and gtalk's user from the ARA database, as I have already done with sip and iax users? I need to know what family I must map to my database (such as iaxusers, sipusers... etc) to enable loading from the database of jabber and gtalk's peers. I also need to know what columns these tables need, there's any documentation online on it? Thank everyone ___ --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] [OT] wi-fi ip phone scenario
Yes, same channel and same ESSID for all AP's. Are you connecting each AP to the LAN? Or only one connected, and the others as relay? With WDS, you have to keep same channel and ESSID for a good roaming. If connected to the lan, doing it worked really good for me, roaming was working in the same way as with WDS (no latency). Le Lundi 30 Octobre 2006 08:31, Alberto Pastore a écrit : Alban ha scritto: I've made some tests with Hitachi WIP3000 and 5000, works really good with roaming (without authentification). Some parts of the AP in the mesh are wired (no WDS), some others are not (using WDS), but all use the same SSID and channel. In all cases roaming was fast, quite not possible to hear it. Besides, with UTstarcom, roaming in the same mesh was not working well. Hope it helps Alban So you're all using access points on the same channel? I've been told that to make a good roaming wifi lan access points must have coverage areas overlapping with the next ap for a good 20%-25% of it (just to let the client roam to another AP while moving), and they should use different channels according their topology, in order to minimize adiacent channel interference, for instance something like this 1--13-4 10-6--9 8--12-2 I'm confused... ___ --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] No ring tone when using IAX
this is really ugly workaround, because using r option in dial you
lose any other progress tones, including busy, congestion, and you will
always hear ring tone even in case of congestion...
PJ
Michiel van Baak wrote:
check your Dial call. You can add a r to the options. That
way it will generate ring tone while waiting for the other
side to pickup.
exten = s,n,Dial(IAX2/sometrunk/${NR_TO_DIAL},45,r)
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[asterisk-users] Intel S3000AHLX - Digium TE110P
Does anybody use Intel S3000AHLX board with Digium TE110P E1 card? Have you experienced any problems? I'm planning following configuration, so I would appreciate any experience both positive and negative. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Linksys PAP2: calling tone stops after 5 tones
Sunday, October 29, 2006, 10:00:22 PM, Jose Limeres wrote: I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? You may have some mistake in dialplan of pap2. I have had the same problem when I put || (two pipe) in dialplan of the line (in advanced admin setting). In this situation the timeout depends on interdigit short timer on regional settings tab. -- Best regards, Csibra Gergomailto:[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
[asterisk-users] Problem with incomming calls
I've got an odd situation where callerid is only picked up every other call. Is there anything I can do so callerid works on all calls?I'm seeing the channel hangs up during a call == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1'Oct 30 10:20:04 NOTICE[31661]: chan_zap.c:5889 ss_thread: Got event 2 (Ring/Answered)...Oct 30 10:20:06 WARNING[31661]: chan_zap.c:5929 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Answer("Zap/4-1", "") in new stack -- Executing NoOp("Zap/4-1", "") in new stack -- Executing LookupCIDName("Zap/4-1", "") in new stackWhen it works I see this: == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' -- Executing Answer("Zap/4-1", "") in new stack -- Executing NoOp("Zap/4-1", "01946622960") in new stack -- Executing LookupCIDName("Zap/4-1", "") in new stackzapata.conf[channels]switchtype=nationalusecallerid=yesukcallerid=yescidsignalling=v23cidstart=polarityhidecallerid=nosendcalleridafter=2callwaiting=yes;answeronpolarityswitch=nousecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesTIAPhil___
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[asterisk-users] anti ex-girlfriend
Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 |2 | hangup | | 455 | DID | 14193016880 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman it's work good. but for routing by caller id: 456 | DID | 14193016880/2085838 |2 | hangup || 455 | DID | 14193016880/2085838 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman this extension does not work , with a call from 2085838 please help me tanx Pezhman We have the perfect Group for you. Check out the handy changes to Yahoo! Groups (http://groups.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Digium 400P and asterisk 1.4
Hi. Ever since I bought my Digium 400P with 1 FXS and 1FXO module, once in a while I hear what sounds like a touchtone in my ear on a phone hooked up to the FXS module. This was not heard by the other side, and although it was annoying, it was not too much of a problem till I was using the asterisk 1.4 (rev 46317) and the beta of zaptel 1.4 (rev 1536). Doing this, the toutchtone noises once heard stay on indefinitely, so I can't hear anything else. If I go back to the 1.2 zaptel then this problem does not occur. Anyone know what is going on here? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] anti ex-girlfriend
Has far as I know, Asterisk doesn't support ex-girlfriend logic in realtime extensions so far. Regards, Ricardo. Pezhman Lali wrote: Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 |2 | hangup | | 455 | DID | 14193016880 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman it's work good. but for routing by caller id: 456 | DID | 14193016880/2085838 |2 | hangup || 455 | DID | 14193016880/2085838 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman this extension does not work , with a call from 2085838 please help me tanx Pezhman We have the perfect Group for you. Check out the handy changes to Yahoo! Groups (http://groups.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RT Problem: Asterisk Session Border Controller
Hi All If this is the incorrect place or people can suggest a better forum to post or an Asterisk consulting service I would be most grateful. I am using an SBC with our * server. All end points register via the SBC to the * server. The sip_buddies table has an entry for user and user . User is able to register and make calls without any problems. As soon as I register user stops being able to make any calls. If I remove user from the RT database and have the entry in the sip.conf reload and re-register then this user will work fine as will user (still running from RT) In my sip.conf I have the following (although I have tried without these): rtcachefriends=yes rtupdate=yes rtautoclear=yes When the users are registered my CLI output shows: (10.10.10.1 = Internal IP of SBC) / 10.10.10.1 D 5060 Unmonitored / 10.10.10.1 D 5060 Unmonitored The Realtime Database shows: fullcontact: sip:[EMAIL PROTECTED]:5060;transport=UDP fullcontact: sip:[EMAIL PROTECTED]:5060;transport=UDP When I try to make a call from phone after has registered I see the invites as I would expect and all looks ok but the * server replies with a 403 Forbidden error. I have tailed the full logfile and I can see the SQL commands RT is using to check the extensions etc but I never see the checks for authentication (i.e that he endpoint has permission to make the call) although I do see this in the packet information) I can see the problem occurs when using RT and when placing calls but with all the information in front of me I cannot deduce why the * server sends back this 403 error. Many Thanks in Advance Scott ___ --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] Vgsm driver 0.18.0 released today
Hi! for those using vGSM cards, today we released version 0.18.0, that fixes a lot of small things and implements a lot of new features, also to improve performances! In the weekend on a test customer the channel driver made 67000+ dials without a glitch! Users are encouraged to upgrade! Please take a look on http://open.voismart.it Cheers, Matteo. -- Come and visit us at VON Italy, Rome From Oct 25 to Oct 26 - Hotel Ergife Matteo Brancaleoni RD Director Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [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
[asterisk-users] Need Help in Meetme (Conferencing)
Hi all,Suppose I have a simple conference configuration as below ---meetme.conf[general][rooms]conf = 0041435215311-and I have a dial plan like this ---extensions.confexten = 0041435215311,1,Answerexten = 0041435215311,2,Wait(1)exten = 0041435215311,3,Agi(agi://localhost/agiconference.agi)exten = 0041435215311,4,MeetMe(0041435215311|p)exten = 0041435215311,5,Playback(vm-goodbye)exten = 0041435215311,6,Hangup-How can I Call someone using Originate Call in Manager API to the conference.If this are the parameters for Manager API call , what should i set in Channel to make a call from conference to the guy who I want to invite, will this work at all?CallerID = 0041435215311Channel = ?Context = defaultExten = "Extension of the one who i want to call him to join conference"Priority = 1Best Ehsan___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Help in Meetme (Conferencing)
Ehsan Khosrowshahi wrote: Hi all, Suppose I have a simple conference configuration as below -- - meetme.conf [general] [rooms] conf = 0041435215311 - and I have a dial plan like this -- - extensions.conf exten = 0041435215311,1,Answer exten = 0041435215311,2,Wait(1) exten = 0041435215311,3,Agi(agi://localhost/agiconference.agi) exten = 0041435215311,4,MeetMe(0041435215311|p) exten = 0041435215311,5,Playback(vm-goodbye) exten = 0041435215311,6,Hangup - How can I Call someone using Originate Call in Manager API to the conference. If this are the parameters for Manager API call , what should i set in Channel to make a call from conference to the guy who I want to invite, will this work at all? CallerID = 0041435215311 Channel = ? Context = default Exten = Extension of the one who i want to call him to join conference Priority = 1 try Channel = SIP/Extension of the one who i want to call him to join conference Context = default Exten = 0041435215311 Priority = 1 Best Ehsan ___ --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] Mac OS X Desktop / Asterisk integration?
Hi, We are successfuly using TAPI with Asterisk in order to provide a generic and fairly well supported interface from Windows desktops to Asterisk - This allows caller-id popping and click-to-dial from TAPI aware environments. Is there an equivalent telephony interface available for Mac OS X, and if so, is there an asterisk-manager plugin for it? Thanks for any feedback - All of the searching I have done has ended in pages on the WiKi telling me how to install Asterisk on a Mac :( Cheers, 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
[asterisk-users] SIP Server
Hi Dear Users, I am new to Asterisk and had a query which is probably primitive. I wanted to know whether I can use the Digium Hardware and receive and establish connection to a host SIP Server which is totally a different platform. Let me explain - Usually there is a E1-VoIP gateway (independent Hardware) connecting to a Server/Client via LAN. In my case, SIP Server. Now what I want is that Digium PCI Hardware and the SIP Server will be the same PC and I Want the PCI Hardware to act as the gateway. Therefore my question in particular is: That is can I configure the device to talk to the Server in SIP protocol directly?-- Imran M Yousuf ___ --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] Realtime trouble with contex
Hello, Asterisk. I am currently using Asterisk (asterisk-1.2.13) and asterisk-addons-1.2.3_1 on FreeBSD 6.1-RELEASE-p10 So, after setup asterisk for realtime extension: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = asterisk dbport = 3306 dbsock = /tmp/mysql.sock res_odbc.conf: [mysql] enabled = yes dsn = asterisk username = asterisk password = asterisk pre-connect = yes extconfig.conf: sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies voicemail = mysql,asterisk,sip_buddies extensions = mysql,asterisk,extensions queues = mysql,asterisk,sip_buddies queue_members = mysql,asterisk,sip_buddies extensions.conf: [office] include = demo ;exten = 333,1, Macro(stdexten,333,SIP/333) ;exten = user1, 1, Goto(333|1) ;exten = 222,1, Macro(stdexten,222,SIP/222) ;exten = user1, 1, Goto(222|1) ;exten = 201,1, Macro(stdexten,201,SIP/201) ;exten = user2, 1, Goto(201|1) ;exten = 202,1, Macro(stdexten,202,SIP/202) ;exten = user3, 1, Goto(202|1) switch = Realtime/[EMAIL PROTECTED] from user 333 i dialing 222 and get this message: Executing Macro(SIP/333-086e2000, stdexten,222,SIP/222) Oct 30 14:29:39 WARNING[65203]: app_macro.c:160 macro_exec: No such context 'macro-stdexten,222,SIP/222' for macro 'stdexten,222,SIP/222' if i uncomments lines: exten = 333,1, Macro(stdexten,333,SIP/333) exten = user1, 1, Goto(333|1) exten = 222,1, Macro(stdexten,222,SIP/222) exten = user1, 1, Goto(222|1) in file extensions.conf, this message absent. in mysql config: mysql select * from extensions; ++-+---+--+---+--+ | id | context | exten | priority | app | appdata | ++-+---+--+---+--+ | 1 | office | 333 |1 | Macro | stdexten,333,SIP/333 | | 2 | office | user1 |1 | Goto | 333|1| | 3 | office | 222 |1 | Macro | stdexten,222,SIP/222 | | 4 | office | user2 |1 | Goto | 222|1| ++-+---+--+---+--+ how to solve this problem? Thanks. -- Senior Systems Engineer Nikita Olenets[EMAIL PROTECTED] ZEON-UANIC ZEON-RIPE ___ --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] Extension Matching with Match As You Go Dialing
Hi all, when calling from the PSTN with Match As You Go Dialing (lift the handset before start to dial) over a zap channel Asterisk simply takes the first extension digit and tries to match it. Because no valid one-digit-extension exists in my dialplan, matching fails and Asterisk says that a invaild extension was dialed. On voip-info.org I read that zap channels have a fixed timeout period of 3 seconds. But in my case Asterisk does not wait even 100 ms before taking the invalid extension. How can I tell Asterisk to wait for more extension digits, especially when the first digit of the dialed extension is no vaild extension number? Thanks in advance. Regards, Jens ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Server
Yes, you just need to setup asterisk with Digium board on the same server of your sipserver, and then you must establish a trunk between your sip server and asterisk. Then you must route calls using asterisk dialplan as well as your sip server dialplan. Be aware that if you have both on same server you must change SIP port in one of them. On 10/30/06, Imran M Yousuf [EMAIL PROTECTED] wrote: Hi Dear Users, I am new to Asterisk and had a query which is probably primitive. I wanted to know whether I can use the Digium Hardware and receive and establish connection to a host SIP Server which is totally a different platform. Let me explain - Usually there is a E1-VoIP gateway (independent Hardware) connecting to a Server/Client via LAN. In my case, SIP Server. Now what I want is that Digium PCI Hardware and the SIP Server will be the same PC and I Want the PCI Hardware to act as the gateway. Therefore my question in particular is: That is can I configure the device to talk to the Server in SIP protocol directly? -- Imran M Yousuf ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] show logged clients
hi all, in console mode how i can display the logged 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] show logged clients
in sip sip show peers (or users) in iax iax2 show peers (or users) if u want more datailled view of the 208 user sip show peer 208 Pablo Allietti a écrit : hi all, in console mode how i can display the logged 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 -- Jean-Baptiste Bellet Ingénieur Développpement Lucyde SAS Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex +33 (0)5 34 31 86 36 http://www.lucyde.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] anti ex-girlfriend
Take a look at smartCID (found at www.generationd.com) You can take actions such as block/limit call times/accept based caller number. It will also fill in the missing CID name based on database lookup (or 411 reverse lookup). MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Monday, October 30, 2006 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] anti ex-girlfriend Has far as I know, Asterisk doesn't support ex-girlfriend logic in realtime extensions so far. Regards, Ricardo. Pezhman Lali wrote: Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 |2 | hangup | | 455 | DID | 14193016880 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman it's work good. but for routing by caller id: 456 | DID | 14193016880/2085838 |2 | hangup || 455 | DID | 14193016880/2085838 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman this extension does not work , with a call from 2085838 please help me tanx Pezhman We have the perfect Group for you. Check out the handy changes to Yahoo! Groups (http://groups.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Mac OS X Desktop / Asterisk integration?
I used some of the ideas found here:http://www.voip-info.org/wiki/view/Asterisk+manager+ExamplesOn 10/30/06, Steve Davies [EMAIL PROTECTED] wrote: Hi,We are successfuly using TAPI with Asterisk in order to provide ageneric and fairly well supported interface from Windows desktops toAsterisk - This allows caller-id popping and click-to-dial from TAPI aware environments.Is there an equivalent telephony interface available for Mac OS X, andif so, is there an asterisk-manager plugin for it?Thanks for any feedback - All of the searching I have done has ended in pages on the WiKi telling me how to install Asterisk on a Mac :(Cheers,Steve___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___ --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] Mac OS X Desktop / Asterisk integration?
On 10/30/06, Tom Vile [EMAIL PROTECTED] wrote: I used some of the ideas found here: http://www.voip-info.org/wiki/view/Asterisk+manager+Examples Okay, so I didn't think to search from the AstManager perspective :) Thanks for the pointer. 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
[asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000
Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk Manager
Dear Michiel and Supporters!
Thank you for your reply!
Here is the complete code for my monitor.php
html
head
titleAsterisk Status/title
link href=common/classic.css rel=stylesheet type=text/css
/head
body
h2Asterisk Status: ? echo `hostname`; echo (.$SERVER_ADDR.); ?/h2
?
$arr = array(
Uptime = asterisk -r -x 'show uptime',
Database Connection Status = asterisk -r -x 'realtime mysql
status',
Active Channel(s) = asterisk -r -x 'sip show channels',
Working Queues = asterisk -r -x 'show queues',
Registered Phones = asterisk -r -x 'sip show peers',
Zaptel driver info = asterisk -r -x 'zap show channels',
);
foreach ($arr as $key = $value) {
?
br
table class=box
tr class=boxheader
td class=boxheader? echo $key; ?/td
/tr
tr class=boxbody
td
table border=0
tr
td
pre? echo passthru($value); ?/pre
/font/td
/tr
/table
/td
/tr
/table
?
}
?
/body
/html
# End of monitor.php file ##
The manager.conf as I have posted previously.
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 25, 2006 1:37 PM
Subject: Re: [asterisk-users] Re: Asterisk Manager
On 13:12, Wed 25 Oct 06, Maps wrote:
Dear Friends and Supporters!
I try to write a php application to monitor the asterisk, but when I
open the .php to access to asterisk to retrieve the information about the
queues status, sip show peers, realtime mysql status etc... However, It
just return to me Unable to connect to remote asterisk (does
/var/run/asterisk.ctl exist?)
Asterisk is current running with the a file in /var/run/asterisk.ctl for
the user asterisk. I have set asterisk to be the owner of the folder
/var/run, and start asterisk with user is asterisk. HTTPD is run under
asterisk user. My manager.conf has an entry.
[admin]
secret = password
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
However, my php still unable to retrieve the information for asterisk.
Did I miss somethings?
How are you connecting to asterisk?
Maybe you can paste some code so we can actually see why it
is not working.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[asterisk-users] Re: IAX2 show peers - description
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi people, pls does anybody know what (T) and (D) letter means? server3*CLI iax2 show peers Name/UsernameHost Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0 unmonitored] Hi Marian, Near host you can have D (dynamic) or S (static). Near port you can have T, but I don't know what it means. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000
Zeeshan Zakaria wrote: Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So I don't specify it on the phone. My Asterisk server changes it's time and all of the phones pick it up. 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] porting numbers in UK telewest/bt/adept
On Sun, Oct 29, 2006 at 03:09:42PM +, Conrad Wood wrote: On 29 Oct 2006, at 11:02, Matthew Thompson wrote: On 26 Oct 2006, at 11:59, Conrad Wood wrote: A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. What is the old PBX and how are Telewest presenting? We had Telewest lines once and they were the same RJ-45 ISDN 30 as BT. Would it not be possible to use a dual port card and use Adept for the outgoing and Telewest for the incoming service? Ah - I forgot to mention that there are 2 offices involved. The client moved to new premises and the telewest lines are in the old office, Adept in the new office. Otherwise I would do as you suggest, yes. Though most OLO's have the ability and can port numbers away from BT, BT will not port numbers out of area i.e. if they are geographic numbers in BT terms they are tied to an exchange. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] How to do Automatic Daylight Saving on Grandstream GXP-2000
Zeeshan Zakaria wrote: Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria And with the change in the US next year, adding three weeks in the Spring and one week in the Fall, where will ALL those smart DST devices be?? John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000
If you have automated the configuration process, all you have to do is:1) Set option P75 (Daylight savings time) to 0 or 1 accordingly in the configuration file.2) Regenerate the compiled configuration file(s).3) Make sure they are in the TFTP or Web server.4) Reboot the phones so they read the new configuration file.- DanielOn Oct 30, 2006, at 9:21 AM, Zeeshan Zakaria wrote:Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks-- Zeeshan A Zakaria ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000
John Novack wrote: And with the change in the US next year, adding three weeks in the Spring and one week in the Fall, where will ALL those smart DST devices be?? I was thinking that this weekend as well. What a waste. 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
[asterisk-users] Re: How to do Automatic Daylight Saving on Grandstream GXP-2000
DL == Doug Lytle [EMAIL PROTECTED] writes: DL I don't specify it on the phone. My Asterisk server changes it's DL time and all of the phones pick it up. The phones get their time from Asterisk? Which protocol do they use for that? /Benny ___ --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] Realtime in the Real World
We are hosting multiple companies with Asterisk. For a high degree of control, each company has many contexts that are included from a main context. I had wanted to use realtime, but realised very soon that it didn't scale. For each context that you put a realtime switch statement in, Asterisk has to go and query the database. If you include 10 contexts, and each one of those has a realtime switch, than that's 10 times that Asterisk has to query the database, for a single call. Combine this with the bizarre behaviour Asterisk has of making the same query several times for a call (why does it do that?) and you have an inordinate amount of queries. I wouldn't like to combine that with several hundred simultaneous calls. Is this really a problem as I see it, or am I missing something? 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] [OT] wi-fi ip phone scenario
Alban ha scritto: Yes, same channel and same ESSID for all AP's. Are you connecting each AP to the LAN? Or only one connected, and the others as relay? With WDS, you have to keep same channel and ESSID for a good roaming. If connected to the lan, doing it worked really good for me, roaming was working in the same way as with WDS (no latency). No WDS, all APs hardwired. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: IAX2 show peers - description
Hi, I think the (T) is for Trunk. Regards Fred ___ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: lundi, 30. octobre 2006 15:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re: IAX2 show peers - description In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi people, pls does anybody know what (T) and (D) letter means? server3*CLI iax2 show peers Name/UsernameHost Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0 unmonitored] Hi Marian, Near host you can have D (dynamic) or S (static). Near port you can have T, but I don't know what it means. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Asterisk and Siemens C450IP
Hi. Again one big mysterious problem I hope some good guy can help me solve. I'm trying to connect some Siemens C450 SIP IP Dect phones to asterisk (1.2.13) (I have actually 3 handsets + 3 ip base). After configuring them and rebooting, all of them register properly on asterisk, then, after the first call, they appear no more registered as registered in asterisk, and on the handset the display shows SIP registration failed. Has anyone got the same problem? Has anyone ever tried to operate more than one C450IP in the same open space? fyi: - there are no lan/ethernet problems - I've tried with different qualify values (0, 2000, 5000) - no nat involved - everything is running on a private lan - phones reply to ping even when qualify times out - any other sip phone (wired to the lan) works just fine It looks like the rise of the machine... every new phone I try drives me crazy. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000
Zeeshan Zakaria wrote: Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria Hi, Firmware 1.1.1.14 for the GXP2000 supports automatic Dalight Savings Time adjustment (AFAIK it was introduced in 1.1.1.11). Upgrade the firmware and add the following to your .cfg file or set via the webpage... # Daylight Savings Time: 0 - No, 1 - Yes P75 = 1 # Daylight Savings Time optional rule P246 = 04,01,7,02,00;10,-1,7,02,00;60 See the Release Notes for the HT386, 488 and 496 at http://www.grandstream.com/DOWNLOAD/FIRMWARE/HT386_488_496/Release_Note_HT386-488-496_1_0_3_44.pdf for details of the Optional Rule regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] How to do Automatic Daylight Saving on Grandstream GXP-2000
Zeeshan Zakaria wrote: Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria Hi, Firmware 1.1.1.14 for the GXP2000 supports automatic Dalight Savings Time adjustment (AFAIK it was introduced in 1.1.1.11). Upgrade the firmware and add the following to your .cfg file or set via the webpage... # Daylight Savings Time: 0 - No, 1 - Yes P75 = 1 # Daylight Savings Time optional rule P246 = 04,01,7,02,00;10,-1,7,02,00;60 This rule is for Eastern Standard Time in the US and Canada, for other locations, see the Release Notes for the HT386, 488 and 496 at http://www.grandstream.com/DOWNLOAD/FIRMWARE/HT386_488_496/Release_Note_HT386-488-496_1_0_3_44.pdf for details of the Optional Rule regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Waiting before executing System command
Hello, Moses!
At 09:20 PM 27/10/2006, you wrote:
what about
exten = h,n,System(mycommand /some/file /some/other/dir/)
Where mycommand is your custom shell script to sleep before moving the file.
That would work, but I'm trying to avoid kludges like that. Hence my
question about doing it entirely within the dialplan.
Any ideas?
On 10/27/06, Alexander Burke [EMAIL PROTECTED] wrote:
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten = h,n,Wait(5)
exten = h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten = h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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[asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme server for 8 more calls. Everything works fine in terms of the asterisk/meetme. The issue arises when the calls comes in via the ATA286 box and in any part of the meeting the CALLER hangs up but the ata286 does not realize the caller hung up so the channels remains open and everyone in the room hears a busy signal. After 30 seconds the ATA286 hangs up (I guess due to timeout) and then the tdm04b hungs the channel and then the meetme room gets back to normal. This is an ATA286 issue right? nothing to do with the TDM or the asterisk box? Since I do not own the ATA286 (the voip provider does) would you recommend something to be asked/changed to the provider of the ATA? Thanks, -- Erick Perez ___ --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] Looking for Wireless Heaset for Polycom 501
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jim Rice Sent: Wednesday, 25 October 2006 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for Wireless Heaset for Polycom 501 We've used the Plantronics CS50 wireless Headset with the HL10 Handset Lifter. About $240. The handset lifter leaves a lot to be desired with the 501. It lifts the handset off the cradle, but doesn't completely hang it up properly. We've had to place items under the phone to tilt it back. Other than that, the headset is great. asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It still amazes me that this isn't built into any handsets yetit seems totally obvious to me to put a wireless component either in or directly connected to (side card connection maybe). This handset lifter policy is just plain crap. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Realtime in the Real World
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG If you include 10 contexts, and each one of those has a realtime
DG switch, than that's 10 times that Asterisk has to query the
DG database, for a single call.
Not that I would make extensions.conf realtime, but...
One trick to avoid includes is to use a Goto multiplexer instead...
Like:
_2XX,1,Goto(internal,$EXTEN,1)
_18005551XXX,1,Goto(movienumbers,${EXTEN},1)
and so on and so forth.
That way you're down to two queries instead of ten.
/Benny
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[asterisk-users] operator console
Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension, and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --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] Looking for Wireless Heaset for Polycom 501
On 10/30/06, Dean Collins [EMAIL PROTECTED] wrote: We've used the Plantronics CS50 wireless Headset with the HL10 Handset Lifter. About $240. The handset lifter leaves a lot to be desired with the 501. It lifts the handset off the cradle, but doesn't completely hang it up properly. We've had to place items under the phone to tilt it back. Other than that, the headset is great. It still amazes me that this isn't built into any handsets yetit seems totally obvious to me to put a wireless component either in or directly connected to (side card connection maybe). This handset lifter policy is just plain crap. I was wondering the same thing. But I'm still not sure if I need the lifters. I connected up the phones but didn't have time to install the lifters. The staff called today and said they have it working without the lifters. So, can someone confirm why I need the lifters? I have to agree that if the lifters are needed, then there should be a phone that comes with this built in. -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How to do Automatic Daylight Saving on Grandstream GXP-2000
Benny Amorsen wrote: all of the phones pick it up. The phones get their time from Asterisk? Which protocol do they use for that? The Polycom's use NTP. And I point the NTP to the Asterisk server. 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
[asterisk-users] Billing Solution ?
Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... Thanks bye ___ --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] Anyone got a dialplan for SPA ATAs for ISN?
After John Todd's talk at Astricon about the ISN project, I spent much of the weekend playing around with it. I have discovered that the default dialplans on my Sipura gear, as well as my Grandstream phones, intercept the * key that is a required part of ISN numbers and interpret it as a metacharacter. Googling for a while has turned up evidence that this can be corrected by a carefully-crafted dialplan for the Sipuras, at least, but the avaialable documentation is, let's say, a little convoluted. I'm wondering if anyone on the list has cracked this, and would be willing to share the gobbledygook string needed to effect the proper behavior. Thanks. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing Solution ?
Try www.asterisk2billing.org Noc Phibee escribió: Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... Thanks bye ___ --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] Asterisk Voicemail with ODBC Realtime Access
Sorry to re-post, but as noone has answeredme ...
Maybe somebody will this time :o)
Thanks !
JM
On 10/29/06, Jean-Marc Salsa wrote:
Hi
I was trying to have realtime voicemail working with ODBC Driver.Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page )
But I do not manage to make it work with ODBC.Outside Asterisk, ODBC works fine, I can access my databases tables !
Asterisk fails to start if I use pre-connect = yes in res_odbc.conf ( See errors below )If I do not use it, then, I get another error message : res_config_odbc.c: SQL Alloc Handle failed! when I try to access the voicemail.
Any ideas would be more than welcome !
Thanks !
Here is my config
/etc/odbcinst.ini[MySQL]Description = ODBC for MySQLDriver = /usr/lib/libmyodbc.soSetup = /usr/lib/libodbcmyS.soFileUsage = 1
/etc/odbc.ini[MySQLast]Description = MySQL ODBC Driver TestingDriver = MySQL#Socket = /var/run/mysqld/mysqld.sockTrace = YesTraceFile = odbc_mysql.logServer = localhost
User = asteriskuserPassword = amp109Database = asteriskrealtime#Option = 3Port = 3306
isql -v MySQLast and then help shows me my Tables correctly,SELECT queries work perfectly
/etc/asterisk/res_odbc.conf[asterisk]enabled = yesdsn = MySQLastusername = asteriskuserpassword = amp109;pre-connect = yes
/etc/asterisk/extconfig.conf[settings]voicemail = odbc,asterisk,voicemail_users
Modules in Asterisk :asterisk*CLI show modules like odbcModule Description Use Countres_config_odbc.so ODBC Configuration 1
res_odbc.so ODBC Resource 0cdr_odbc.so ODBC CDR Backend 03 modules loaded
Error Messages:
if pre-connect = yes is used in res_odbc.confOct 29 16:23:21 VERBOSE[15016] logger.c: MySQL RealTime driver loaded.Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_adsi.so]Oct 29 16:23:21 VERBOSE[15016]
logger.c: [res_adsi.so] = (ADSI Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so]Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so] = (ODBC Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c
: == Parsing '/etc/asterisk/res_odbc.conf': Oct 29 16:23:21 VERBOSE[15016] logger.c: == Parsing '/etc/asterisk/res_odbc.conf': FoundOct 29 16:23:21 NOTICE[15016] res_odbc.c: registered database handle 'asterisk' dsn-[MySQLast]
Oct 29 16:23:21 NOTICE[15016] res_odbc.c: Connecting asterisk
And this is what I get on the standard error output:*** glibc detected *** malloc(): memory corruption: 0x09f4daf0 ***/usr/sbin/safe_asterisk: line 50: 14920 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.*** glibc detected *** malloc(): memory corruption: 0x0a023af0 ***/usr/sbin/safe_asterisk: line 50: 15016 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.
If no pre-connect, then I can start asterisk, but I get this error when I try to access VM:Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing VoiceMail(IAX2/telegrupp-4,
[EMAIL PROTECTED]|) in new stackOct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'
Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4' Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'Oct 29 16:24:43 WARNING[15262] res_config_odbc.c: SQL Alloc Handle failed!
Oct 29 16:24:43 WARNING[15262] app_voicemail.c: No entry in voicemail config file for '200' Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Goto(IAX2/telegrupp-4, exit-FAILED|1) in new stack
Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Goto (macro-vm,exit-FAILED,1)Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Playback(IAX2/telegrupp-4, im-sorryan-error-has-occured) in
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Re: [asterisk-users] Billing Solution ?
On Mon, 2006-10-30 at 18:31 +0100, Noc Phibee wrote: Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... I'm using a2billing and works like a charm for me :) Regards, Thanks bye ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Configuring 2 Asterisk servers with a SIP trunk
Asterisk B: Create an extension (just as you would if you want to connect a SIP client)Asterisk A: Have this guy register using the extension (just as you would using a SIP client like SJPhone) - You will probably have to use type=peer though. Make sure you take care of NAT and stuff like that if neededrajeevOn 10/28/06, Alok Mohapatra [EMAIL PROTECTED] wrote: Hi All, Please let me know the how to configure a SIP trunk of a asterisk Server with another one (not IAX2). Asterisk-A should register a SIP trunk with Asterisk-B server . With Regards Alok Ranjan Mohapatra Software Engineer +91 9866269992 PrimeSoft IP Solutions (P) Ltd # 917- 922,East Wing, 9th floor Block III, White House,Begumpet Hyderabad - 500016, INDIA Ph - 91-40-23418239/40 www.primesoftindia.com ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] operator console
...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? Have a look at FOP : http://www.asternic.org/ ___ --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] Wildcard X100P Suport
Is the Wildcard X100P still supported? I have one sitting around that I bought 3+ years ago and never used it. I need the functionality now. Before I run off and buy something new, I'm curious if this will just work. I also have an old TDM400P with 2 FXS modules that I bought at the same time. Then, there was no FXO module for the 400P. Will a TDM400P this old support a new X100M? Or am I just better off getting everything new? Thanks, MikeC ___ --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] Incorrect Ring tone. Getting a US tone when itshould be AU tone
Hi David, It can be set on the Sipura / Linksys devices. Look under Admin, Advanced, Regional, Call Progress Tones. There is a link floating around on Whirlpool forums to a page and auto provision file containing the correct settings to produce Australian tones. It also depends on whether the phone allows the PBX to make the progress tones or whether the phone alone does them. The setting to control this is in Sipura/Linksys firmware 3.1.10 and higher from memory. If you want the handset to do the tones, or you get a 'double ringing' in the handsets of these phones / ATA's then set Admin / Advanced / Line X / SIP Settings / Sticky 183 to no. Btw, how's your Asterisk going? I'm in the middle of doing a 7 site Least Cost Routed DUNDi setup with redundant routes - Good fun though the learning curve is a bit steep. Craig - Original Message - From: Klaverstyn, David C [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:06 PM Subject: RE: [asterisk-users] Incorrect Ring tone. Getting a US tone when itshould be AU tone I don't think it is a phone problem. I get a US ring tone on a PAP2, SPA-942 and IDEFdisk softphone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Monday, 30 October 2006 5:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incorrect Ring tone. Getting a US tone whenit should be AU tone What phones are you using? It could be a phone level issue. (my aastra has a setting for AU sounds..) PaulH On Mon, 2006-10-30 at 16:13 +1100, Klaverstyn, David C wrote: For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general] country=au ___ --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
[asterisk-users] Asterisk and Panasonic KX Model
If Someone did that, How I connect extensions.conf with this type of Hybrid system to work with asterisk inside this schema: PSTN---PANASONIC KX -- Asterisk | |-send internal call Thanks. ___ --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] Audiocodes MP-20x
hican you please post some user or config guide.thanks in advancearunOn 10/24/06, Ed Greenberg [EMAIL PROTECTED] wrote:I will sign in with good experiences with MP124 and Mediant 1000. I have an MP202 under test.--On Tuesday, October 24, 2006 10:10 AM +0300 Paul Ianas[EMAIL PROTECTED] wrote: I have used AudioCodes MP 102, 104 and 108, both FXS and FXO. I have also used AudioCodes Mediant 2000. I can tell you that these are good devices. There are also many other media gateways that have a lot of facilities, but many of these implement those facilities in software. AudioCodes has also a quite good – let's say -- hardware support. I haven't used MP20x. -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone: 0744137020 email: [EMAIL PROTECTED] __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Andrew Joakimsen Sent: Monday, October 23, 2006 1:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Audiocodes MP-20x Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them...___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Billing Plataforms
Hi, before you start throwing shoes to me, i know there are a lot of Asterisk Billing plataforms, but actually no one seems to accomplish what i need. They are to complex (a2billing) or doesn't have too much documentation (astbill and mcc) or are poorly developed (trabas). What i was looking for is a simple Asterisk billing plataform (with web based admin and customer interface) that only calculates the time of a call and the cost. For example let's suppose that a local extension calls to a UK number. What i need to know is the duration, and the total money spent on that call based on a dynamic tariff DB. I'd like the plataform to use an AGI script because it's a kind of postpaid/prepaid system. There shouldn't be any kind of authentification when the number is dialed. The Account number will be passed to the AGI as a parameter ( AGI(agiscript.agi|account) ) Somebody uses or used or is aware of something like this? Regards, Santiago ___ --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] Good phones for outside of the office?
I am looking for phones that work well (or at all) when outside of the network and behind a router, such as at someone's home or in a hotel. My Polycom IP601s do not seem to be up to the task, so I am hoping that there is a good alternative for my outside sales people to use to talk to my asterisk server. Thanks in advance, Warren ___ --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] TE110P Card
Hi Groupies,I am sort of new to the whole asterisk thing, especially when it comes to the Digium TE110P card. Does anyone have experience setting this up? If so can you help me out? The provider for the PRI is going to be ATT/SBC.ThanksJulian ___ --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] Billing Solution ?
Thanks all for your answer ;=) i start test this week a2billing Noc Phibee a écrit : Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... Thanks bye ___ ___ --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] Fxo box for asterisk ?
Hi do you know if they have external Box (not internal card) for connect Analog Line and Pri/Isdn to asterisk for incomming and outgoing calls ... Thanks ___ --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] light web user interface
Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Forwarding recorded calls to Voicemail
I was wondering if anyone has implemented a feature that would allow a user to record a phone call and once the call has ended, the call is forwarded to his voicemail?ThanksTom Vile ___ --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] Live creation of trunk groups
Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne ___ --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] Wildcard X100P Suport
Is the Wildcard X100P still supported? I have one sitting around that I bought 3+ years ago and never used it. I need the functionality now. Before I run off and buy something new, I'm curious if this will just work. It still works with the latest Zaptel (1.2.10) I also have an old TDM400P with 2 FXS modules that I bought at the same time. Then, there was no FXO module for the 400P. Will a TDM400P this old support a new X100M? I don't think that something changed, so it should work. You should contact Digium to be 100% shure hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Panasonic KX Model
Use an analog extension port on the Panasonic to a FXO port on Asterisk. On 10/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If Someone did that, How I connect extensions.conf with this type of Hybrid system to work with asterisk inside this schema: PSTN---PANASONIC KX -- Asterisk | |-send internal call Thanks. ___ --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] MFC/R2 patch problems
I have looked on this list but may have missed it. I am having problems patching the makefile in the asterisk channels source tree. I keep getting: -- patching file Makefile Hunk #1 FAILED at 72. Hunk #2 FAILED at 143. Hunk #3 FAILED at 178. -- Any ideas for solutions? Thanks, Christian Jensen P.S. Sorry If i am repeating this question. ___ --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] porting numbers in UK telewest/bt/adept
On 29 Oct 2006, at 11:02, Matthew Thompson wrote: On 26 Oct 2006, at 11:59, Conrad Wood wrote: A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. What is the old PBX and how are Telewest presenting? We had Telewest lines once and they were the same RJ-45 ISDN 30 as BT. Would it not be possible to use a dual port card and use Adept for the outgoing and Telewest for the incoming service? Ah - I forgot to mention that there are 2 offices involved. The client moved to new premises and the telewest lines are in the old office, Adept in the new office. Otherwise I would do as you suggest, yes. Conrad ___ --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] Live creation of trunk groups
My advice is to first make some tests to see if a reload is enough for Asterisk to read any group definitions change in zapata.conf, otherwise no on-the-fly change will workAlyed Return-Path: [EMAIL PROTECTED] Mon Oct 30 13:23:36 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;Mon, 30 Oct 2006 13:23:36 -0700 Hi,Is there a way to create trunk groups while asterisk is running.For exemple let's say that zapata.conf defines g0 as channels 1-23I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23Any hints appreciated.Andre Courchesne___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
re: [asterisk-users] Good phones for outside of the office?
Isn't your problem more about NAT traversal rather than the phones themselves?if so better use some iax softphone, have a look at: http://www.voip-info.org/wiki-VOIP+Phonesof course you can use SIP based hard/soft phones but using iax based ones is cheaper and faster.Alyed Return-Path: [EMAIL PROTECTED] Mon Oct 30 12:30:21 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;Mon, 30 Oct 2006 12:30:21 -0700 I am looking for phones that work well (or at all) when outside of the network and behind a router, such as at someone's home or in a hotel. My Polycom IP601s do not seem to be up to the task, so I am hoping that there is a good alternative for my outside sales people to use to talk to my asterisk server.Thanks in advance,Warren___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fxo box for asterisk ?
Check out the SPA-3000 from Sipura (www.sipura.com). On 10/30/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi do you know if they have external Box (not internal card) for connect Analog Line and Pri/Isdn to asterisk for incomming and outgoing calls ... Thanks ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Recommendations
We have a number of clients who will be needing a server to host Asterisk on. Many of these clients use analog (FXO) lines that will need to be connected to Asterisk via Sangoma cards. Can anyone recommend an industry-standard server (like IBM, Dell, HP, etc.) that has enough open PCI slots to handle up to six of the Sangoma cards? We would like to be able to tell the customer to just go purchase this model server from this manufacturer and it will work. Suggestions? Thank you! Joe Dennick ___ --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] Asterisk architecure
Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy. I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? Thank you _ Be the first to hear what's new at MSN - sign up to our free newsletters! http://www.msn.co.uk/newsletters ___ --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] Server Recommendations
How many analog lines are you looking at? Hundreds? PaulH On Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote: We have a number of clients who will be needing a server to host Asterisk on. Many of these clients use analog (FXO) lines that will need to be connected to Asterisk via Sangoma cards. Can anyone recommend an industry-standard server (like IBM, Dell, HP, etc.) that has enough open PCI slots to handle up to six of the Sangoma cards? We would like to be able to tell the customer to just go purchase this model server from this manufacturer and it will work. Suggestions? Thank you! Joe Dennick ___ --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] MFC/R2 patch problems
Learn how a patchfile/Makefile works, and fix the patch. Actually the Makefile patch never has applied cleanly in my experience, so always a few fixes are needed. On 10/30/06, Christian Jensen [EMAIL PROTECTED] wrote: I have looked on this list but may have missed it. I am having problems patching the makefile in the asterisk channels source tree. I keep getting: -- patching file Makefile Hunk #1 FAILED at 72. Hunk #2 FAILED at 143. Hunk #3 FAILED at 178. -- Any ideas for solutions? Thanks, Christian Jensen P.S. Sorry If i am repeating this question. ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] IVR
Greetings, If somebody knows how to concatenate several .gsm files in one or create a macro and use with background() please reply. ___ --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] Incorrect Ring tone. Getting a US tone when itshould be AU tone
Sounds like some nice work - we are currently finishing off a 200 seat call centrewhich has just been _hard_ work. PaulH On Tue, 2006-10-31 at 02:32 +0800, Craig Guy wrote: Hi David, It can be set on the Sipura / Linksys devices. Look under Admin, Advanced, Regional, Call Progress Tones. There is a link floating around on Whirlpool forums to a page and auto provision file containing the correct settings to produce Australian tones. It also depends on whether the phone allows the PBX to make the progress tones or whether the phone alone does them. The setting to control this is in Sipura/Linksys firmware 3.1.10 and higher from memory. If you want the handset to do the tones, or you get a 'double ringing' in the handsets of these phones / ATA's then set Admin / Advanced / Line X / SIP Settings / Sticky 183 to no. Btw, how's your Asterisk going? I'm in the middle of doing a 7 site Least Cost Routed DUNDi setup with redundant routes - Good fun though the learning curve is a bit steep. Craig ___ --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] Architecture for Asterisk
Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy? (or only as a register server?) I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? 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] IVR
exten= X,1,Background(nameoffile) exten= X,2,Background(nameoffile1) For a macro, you need to be passing arguments... what are you trying to pass? You can also use audacity to concat .gsm files but you have to import it from raw data. You will then have to downsample it from 41000 to 8000 hz. -Christian Jensen On Oct 30, 2006, at 6:54 PM, Vitalie Apostu wrote: Greetings, If somebody knows how to concatenate several .gsm files in one or create a macro and use with background() please reply. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and Panasonic KX Model
I published a pretty detailed how to about this on the voip-info.org wiki a couple years ago ... A lot of things have changed since then and I suspect that some of the methods I used may be obsolete at this point ... But I think it might be a good place for you to start ... With my setup it was possible for Panasonic phones to call sip phones and sip phones to call Panasonic phones ... This integration was not perfect but worked well enough for me until I could afford to replace the Panasonic KSU and proprietary phones with all sip phones ... Check out http://www.voip-info.org/wiki/view/Panasonic+KSU and see if this answers some of your questions ... And if you re-write it so it works better than the one I did, please publish your results so others can benefit from you efforts ... G.Hendershot -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, October 30, 2006 1:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Panasonic KX Model If Someone did that, How I connect extensions.conf with this type of Hybrid system to work with asterisk inside this schema: PSTN---PANASONIC KX -- Asterisk | |-send internal call Thanks. ___ --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] Registration problem
Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 5504 REGISTER Max-Forwards: 70 Contact: SPA922 sip:[EMAIL PROTECTED]:1025;expires=3600 User-Agent: Linksys/SPA942-4.1.12 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to x.x.x.x : 1025 (NAT) Transmitting (NAT) to x.x.x.x:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc;received=x.x.x.x From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 5504 REGISTER User-Agent: incore-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce Call-ID: [EMAIL PROTECTED] CSeq: 5503 REGISTER User-Agent: incore-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479 Content-Length: 0 Why the phone can not register? The password and username are ok. Thanks ___ --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] Architecture for Asterisk
Something like this? PaulH On Mon, 2006-10-30 at 16:08 -0800, je . wrote: Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy? (or only as a register server?) I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? 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 attachment: asterisk.jpeg ___ --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] Diva server 4bri and Portuguese BRI
Hello, One problem is solved and another appears... :( I cannot receive incoming calls on trixbox. I defined one incoming route (any DID/any CID) and forwading these calls to a SIP extension. With capi and sip debug in asterisk -r console i dont detect any incoming activity... In xlog console i have the following debug: 0:1898:127 - SIG-R(034) 08 01 0D 05 A1 04 03 80 90 A3 18 01 81 6C 0B 00 83 39 36 33 30 34 35 37 32 33 70 02 81 30 7D 02 91 81 Q.931 CR0d SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 81 Calling Party Number 00 83 '963045723' Called Party Number 81 '0' HLC 91 81 0:1898:127 - SIG-S 0-6 e:805 0:1898:130 - CREATEID ok: context:1f assigned Id:3 freeIds=ec 0:1898:130 - alloc cr in use =4 0:1898:133 - SIG-X(008) 08 01 8D 45 08 02 80 95 Q.931 CR8d DISC Cause 80 95 'Call rejected' 0:1898:133 - SIG-x(008) 08 01 8D 5A 08 02 80 D8 Q.931 CR8d REL_COM Cause 80 d8 'Incompatible destination' 0:1898:133 - SIG-S 6-0 e:8c5 0:1898:134 - D-X(012) 00 01 14 16 08 01 8D 5A 08 02 80 D8 0:1898:135 - free cr in use =3 0:1898:135 - DELETEID ok: deleted Id:4 freeIds=ec 0:1898:155 - D-R(004) 00 01 01 16 So the problem appears to be Incompatible destination... but is problem in asterisk or is before asterisk, on diva card...? Tanks by any possible help! Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Finally this works!!! :) Tanks to Alberto and Marco by your help! The problems are: - the cable was connected to the wong card port... :( - the card config needs to be: ETSI; TE; Point-to-Point (I thought that was point-to-multipoint). Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --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: Architecture for Asterisk
Thanks for the diagram. Is it possible to get a more detailed diagram. I'm looking for something a little more technical. In other words, where does Asterisk stand when inviting a user, when hanging up, when canceling an invitation etc.. Does it go direct from user to user or does it go to Asterisk?JezSomething like this?PaulHOn Mon, 2006-10-30 at 16:08 -0800, je . wrote: Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy? (or only as a register server?) I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? 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-- next part --A non-text attachment was scrubbed...Name: asterisk.jpegType: image/jpegSize: 5585 bytesDesc: not availableUrl : http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/5e764423/asterisk-0001.jpeg___ --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] sip trunk - SIP/2.0 488 Not Acceptable Media
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I'm working on sip trunk between cme 4.x and asterisk (trixbox 1.2.3). Well, the trunk is partially working, asterisk' extensions talk with cme, but - - when cme try to connect to asterisk' number, receives the number dialed is not in service. - - calls from ISP through asterisk to cme don't work completely, and my sip debug says: SIP/2.0 488 Not Acceptable Media even if calls from ISP to asterisk use the same codec as asterisk with cme uses (g711alaw). Any advice to point my troubleshooting in the right direction will be appreciated Best Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFFRpxzMakHrsrHP9wRAsvwAKC2YroRXxpaeqCX3ng+HliAtpPDTACfdX92 ioaAFclJnbRLJg3Y4Eh5duI= =EdFI -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Architecture for Asterisk
Have you had a read of the Asterisk:Future of telephony book? http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 PaulH On Mon, 2006-10-30 at 16:37 -0800, je . wrote: Thanks for the diagram. Is it possible to get a more detailed diagram. I'm looking for something a little more technical. In other words, where does Asterisk stand when inviting a user, when hanging up, when canceling an invitation etc.. Does it go direct from user to user or does it go to Asterisk? Jez Something like this? PaulH On Mon, 2006-10-30 at 16:08 -0800, je . wrote: Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy? (or only as a register server?) I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? 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 -- next part -- A non-text attachment was scrubbed... Name: asterisk.jpeg Type: image/jpeg Size: 5585 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/5e764423/asterisk-0001.jpeg ___ --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] dealing with blind transfers to invalid extensions
Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
What is the best way to do this? Thanks in advance
Here's a snippet of my extensions.conf
[default]
exten=_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten=_11XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
include=record
include=parkedcalls
include=voicepulseoutgoing
include=conferences
include=voicemail
[macro-stdexten]
exten=s,1,Dial(${ARG2},20,t)
exten=s,2,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(u${ARG1})
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(b${ARG1})
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
==
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Re: [asterisk-users] dealing with blind transfers to invalid extensions
Jeronimo Romero wrote:
Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
I do the following:
[direct-to-voicemail]
; **
; Allow anybody to send a call directly to voicemail
; by pre-pending a 0 to the destination extension.
; Checks to see if voice mail box exists, if not
; Tells the callee that no such vm box exists and
; then transfers them to the operator
; **
exten = _04XXX,1,Set(_direct_vm=${EXTEN:1})
exten = _04XXX,2,MailboxExists([EMAIL PROTECTED])
exten = _04XXX,3,Goto(s-${VMBOXEXISTSSTATUS},1)
exten = s-FAILED,1,SayDigits(${direct_vm})
exten = s-FAILED,2,Playback(vm-nobox)
exten = s-FAILED,3,Playback(pbx-transfer)
exten = s-FAILED,4,Goto(incoming,s,1)
exten = s-SUCCESS,1,Set(CALLBACK=${DB(vmcallback/${direct_vm})})
exten = s-SUCCESS,2,GotoIf($[${CALLBACK} =
YES]?s-SUCCESS,3:s-SUCCESS,4)
exten = s-SUCCESS,3,System(/usr/local/bin/vm-callout.sh ${direct_vm})
exten = s-SUCCESS,4,Voicemail([EMAIL PROTECTED])
exten = s-SUCCESS,5,Hangup()
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Re: How to do Automatic Daylight Saving on Grandstream GXP-2000
I already have P246 = 04,01,7,02,00;10,-1,7,02,00;60 value set. I do have TFTP server and the phones read configuration from there when bootup. Also I have: # Daylight Savings Time: 0 - No, 1 - YesP75 = 1 All the phones have the latest firmware. But still they don't do automatic daylight saving. ___ --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] Cisco 7960 Skinny calling SIP phone
Before I got down the path of converting a Cisco 7960 I haveover to SIP I wanted to try and set it up using Skinny. Thephone registersok withAsterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connectedI see the following messag: Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :) regards Wil ___ --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] CID and CDR conflict?
Well, options b is unacceptable since I may be supplied a caller id name, which I would want to pass on to my users. Options a is a bit of a kludge, but I think I could make it work. I'd have a cron job that updates the cdr.src field based on the cdr.user field. That could work but I was sure hoping there was a better way. Thanx for your time. Mike Diehl. On Sunday 29 October 2006 23:01, Leo Ann Boon wrote: Mike Diehl wrote: Hi all, I've been beating my head against this for some time now. For incoming calls, I'd like to send my users a localized caller id number. By localized, I mean one with out the 1+areacode for local calls and only 10 digits (minus the leading 1) for long distance calls. For example: I get a call from 1501234. Since I live in the 505, I should see: 5551234 on my caller id. However, if I get a call from 18035556789, I should see: 8035556789 on my caller id. The problem is that I also want to preserve the original calling number in my CDR(src) field. But everytime I change the CID number, it changes the CDR(src) field. Is there any way I can have the best of both worlds? More than 1 ways to skin the rabbit: a. Set CALLERID(num) to localized number for and store the original cid in the CDR(userfield) b. Set CALLERID(name) to localized number. This method depends on the phone. Some phone will display only the name part if it's supplied (what you want), others will show both. The CDR(src) is preserved but CDR(clid) will look like 5551234 1505551234 Leo ___ --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] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
Erick Perez wrote: PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme server for 8 more calls. Everything works fine in terms of the asterisk/meetme. The issue arises when the calls comes in via the ATA286 box and in any part of the meeting the CALLER hangs up but the ata286 does not realize the caller hung up so the channels remains open and everyone in the room hears a busy signal. After 30 seconds the ATA286 hangs up (I guess due to timeout) and then the tdm04b hungs the channel and then the meetme room gets back to normal. The ATA will be getting the hangup - it'll be what's generating the busy tone you hear when the SIP session between the ATA and your VoIP provider is terminated. If you can get your provider to enable the P205 Polarity Reversal setting on the ATA, the ATA will reverse the polarity of the voltage on it's FXS port when and outgoing call is answered (outbound calls), and when the remote end hangs up (for calls in either direction). You'll then be able to set hanguponpolarityswitch=yes in zapata.conf, and hangups should then be detected almost immediately (with luck, before any tones are heard). HTH, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --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] Audiocodes MP-114 noise
It's noisy while talking. Any idea? Thanks in advance. Jason Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Architecture for Asterisk
Thank you for the link. Chapter 8 was most useful in explaining the different types of connections (user/peer/friend) as well as the register function such that users may know how to contact another user. However I'm looking for something more specific. For instance, for a normal session termination (i.e. BYE), user1 would send msg BYE to proxy who would then forward it to user2 who would then close the connection. For a cancel request the following messages are exchanged: u1 - proxy: invite proxy - u2: arp request u2 - proxy: arp response proxy - u2: invite proxy - u1: trying u2 - proxy: ringing proxy - u1: ringing u2 - proxy: ok proxy-u1:ok u1 - proxy: ack proxy - u2: ack exchange data u2 - proxy: bye proxy - u1: bye In this scenario for instance (where user 2 closes the connection), where does Asterisk fit in? If at all? Does Asterisk behave as the proxy? Thanks, jez - Original Message From: Paul Hales [EMAIL PROTECTED]To: je . [EMAIL PROTECTED]Sent: Monday, October 30, 2006 6:43:13 PMSubject: [Fwd: Re: [asterisk-users] Architecture for Asterisk] Something like this?PaulHOn Mon, 2006-10-30 at 16:08 -0800, je . wrote: Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy? (or only as a register server?) I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? 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
[asterisk-users] Re: [OT] wi-fi ip phone scenario
On 2006-10-29 01:35:46 -0800, Alberto Pastore [EMAIL PROTECTED] said: Martin Joseph wrote: I think it's cleary true that wiring WIFI infrastructure is easier and more reliable then WDS. On the other hand, I have been running my little network with WDS for over three weeks now, and it has been completely reliable. The tricks where to configure things properly and to have the bases closer together then one would think would be needed. Once this was setup. It works, and it keeps working. We had a couple of stress tests also, one black out and one unplugged router (carpenter). Came up cleanly and continued working fine. No mis-registrations and no problems. Marty Can I ask you guys which phones are you using? Nokia E60. It works, but I can see where the next rev. will be better... 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] Cisco 7960 Skinny calling SIP phone
Well, I've never actually been able to make chan_skinny work with 79xx
phones.
I found the chan_sccp to work quite well:
http://chan-sccp.berlios.de/
plus this patch for a problem on MeetMe (I don't remeber where I found
it, but it works!):
diff -uNr chan_sccp-20060408.org/sccp_pbx.c chan_sccp-20060408/sccp_pbx.c
--- chan_sccp-20060408.org/sccp_pbx.c 2006-04-08 14:20:17.0 +0200
+++ chan_sccp-20060408/sccp_pbx.c 2006-05-17 17:14:15.0 +0200
@@ -290,6 +290,12 @@
static int sccp_pbx_answer(struct ast_channel *ast) {
sccp_channel_t * c = CS_AST_CHANNEL_PVT(ast);
+ // if channel type is undefined, set to SCCP
+ if (!ast-type) {
+ sccp_log(1)(VERBOSE_PREFIX_3 SCCP: Channel type
undefined, sett
ing to type 'SCCP'\n);
+ ast-type = SCCP;
+ }
+
if (!c || !c-device || !c-line) {
ast_log(LOG_ERROR, SCCP: Answered %s but no SCCP
channel\n, as
t-name);
return -1;
I recommend using SIP firmware anyway... the conversion process is a bit
annoying but
as far as now 7940/7960 are really stable IP phones.
I am currently using chan_sccp only for 7902 phones (I've just got 2 of
them)
which do not support SIP firmware.
Will Roy ha scritto:
Before I got down the path of converting a Cisco 7960 I have over to
SIP I wanted to try and set it up using Skinny.
The phone registers ok with Asterisk. When I call a SIP softphone
extension on my network the call is made and I can answering it.
However no voice is heard over the call.
When I debug Skinny on the console after the call has connected I see
the following messag:
Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7]
What additional information would be required to troubleshoot this? or
should I stop wasting time and just convert the phone to SIP? :)
regards
Wil
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--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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Re: [asterisk-users] Architecture for Asterisk
je . wrote: Thank you for the link. Chapter 8 was most useful in explaining the different types of connections (user/peer/friend) as well as the register function such that users may know how to contact another user. However I'm looking for something more specific. For instance, for a normal session termination (i.e. BYE), user1 would send msg BYE to proxy who would then forward it to user2 who would then close the connection. For a cancel request the following messages are exchanged: Short answer: Asterisk is an endpoint not a proxy. Leo ___ --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] S(x) - Hang up the call after 'x' seconds - Not working from queue
Hi, I have a requirement to limit the calls to our agents via a queue to 5 minutes. I had posted this to a previous thread by name Maximum talktime in a queue? One work around that was suggested was to use the S(x) in the dial command to the agents, so that all calls to that extension would be terminated after x seconds. So I modified the dial command to the agent as: exten = 1001,1,Dial(SIP/1001,,tS(30)) Now when I call 1001 from another sip phone in the same context it get's disconnected after 30 seconds. -- Executing Dial(SIP/1002-b119, SIP/1001||tS(30)) in new stack -- Setting call duration limit to 30 seconds. -- Called 1001 -- SIP/1001-b605 is ringing -- SIP/1001-b605 answered SIP/1002-b119 -- Attempting native bridge of SIP/1002-b119 and SIP/1001-b605 == Spawn extension (from-sip, 1001, 1) exited non-zero on 'SIP/1002-b119' All is fine so far and it works as advertised. Now I am attempting a call via queue: -- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack -- Started music on hold, class 'default', on channel 'SIP/1002-74e9' -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1001||tS(30)) in new stack -- Setting call duration limit to 30 seconds. -- Called 1001 -- Called Agent/1001 -- SIP/1001-d43c is ringing -- Agent/1001 is ringing -- SIP/1001-d43c answered Local/[EMAIL PROTECTED],2 -- Agent/1001 answered SIP/1002-74e9 -- Stopped music on hold on SIP/1002-74e9 == Spawn extension (from-sip, 1001, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (from-sip, 99, 1) exited non-zero on 'SIP/1002-74e9' This call does not terminate after 30 seconds. I hope I have currently followed the tip from Lenz in my previous tip. with warm regards, raj ___ --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
