Re: [asterisk-users] Skype For Asterisk (SFA)
Dear Abdul Basit, http://nerdvittles.com/index.php?p=784 works, I tested it few months back and it works. Cant say if its still working or not. On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards, Umair Bari -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] timeout with outbound calls
You may also use
exten = 223,n,Dial(SIP/${EXTEN},,KktL(25000))
On Fri, Jul 8, 2011 at 5:33 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
salaheddine elharit
Sent: Friday, July 08, 2011 6:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] timeout with outbound calls
Hi
i want to use timeout with asterisk 1.4 in order to hangup
the outbound calls after 25 sec
i call my mobile number 067xxx from my sip acount 223
and i want to hangu up the call automatic after 25 sec but
there is no hangup after 25
could you please help me
exten = 223,1,Set(TIMEOUT(absolute)=25) exten =
223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
Best Regards.
pbx*CLI core show application dial
-= Info about application 'Dial' =-
[Synopsis]
Attempt to connect to another device or endpoint and bridge the call.
[snip]
L(x[:y[:z]]):
x - Maximum call time, in milliseconds
y - Warning time, in milliseconds
z - Repeat time, in milliseconds
Limit the call to x milliseconds. Play a warning when y mill
iseconds are left. Repeat the warning every z milliseconds until time
expires.
This option is affected by the following variables:
${LIMIT_PLAYAUDIO_CALLER}:
yes
no
If set, this variable causes Asterisk to play the
prompts to the caller.
${LIMIT_PLAYAUDIO_CALLEE}:
yes
no
If set, this variable causes Asterisk to play the
prompts to the callee.
${LIMIT_TIMEOUT_FILE}:
filename
If specified, filename specifies the sound prompt
to play when the timeout is reached. If not set, the time
remaining
will be announced.
${LIMIT_CONNECT_FILE}:
filename
If specified, filename specifies the sound prompt
to play when the call begins. If not set, the time remaining
will
be announced.
${LIMIT_WARNING_FILE}:
filename
If specified, filename specifies the sound prompt
to play as a warning when time x is reached. If not set, the
time remaining will be announced.
[snip]
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks Regards,
Umair Bari
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need suggestion
Dear All, my client wants a asterisk pbx with 30 FXO 30 FXS analogue ports, please suggest if sangoma A400 is a good option for that. Also please suggest server hardware. regards, Umair ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google acquires Grand Central
don't scare people for GOD sake :) LOL On 7/4/07, Jaswinder Singh [EMAIL PROTECTED] wrote: Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . On 03/07/07, Dean Collins [EMAIL PROTECTED] wrote: Ooops did Google just become a carrier :) http://googleblog.blogspot.com/2007/07/all-aboard.html I hear stocks crumbling worldwide as I type. Cheers, Dean ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk management interface
Hello, Try http://www.freepbx.org, its written in PHP with mysql at the end, it also uses .conf files for configurations. regards, Umair bari On 5/8/06, moona ather [EMAIL PROTECTED] wrote: Hi,I have to make a web-based management interface of configuring asteriski wanted to know if it is as simple as reading the .conf files and searching for the required section in the file and adding users etc. or there areother steps involved too?? As I have seen many such built codes on this siteand found lots of code... kindly tell me how complex it is and how many other steps are involved in making this interface as i am new in this.Emmo._Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/___--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] regarding freepbx
freepbx has been improved since then, and I believe if you edit/add something in original asterisk .conf files, it stays there. I've tried it long ago when it was called AMP and it worked. regards, Umair Bari On 5/9/06, Emmo ather [EMAIL PROTECTED] wrote: Hello,In older version of freebpx if you write somethng manually in theconfiguration files it was flushed by amp, i.e. you can configure it throughthe interface only. Is this this thing still present in freepbx?_Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/___--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] Please Help Me...Urgent
Hello, IMHO, there are 2 ways to do this, 1) You can connect your VoIP modem to your asterisk box using x100p FXO card, you'll need to get one and install it properly. 2) Get SIP/IAX account from any VoIP provider and use it with asterisk. Hope this helps. Regards, Umair Bari On 5/12/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi Friends,Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to implement pure VoIP solution. Am I right? I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called Voice Finder AP 200 and the below values: Inbound Number: 123456789Public IP Number: 55.23.789.145Password: xyz(These values are dummy values) Currently we are making US calls using VoIP provided by Vebtel. Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this? I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel? Waiting for your quick response. Thank you. Regards,Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Calls between Asterisk servers using SIP? What about IAX (got it working w/ IAX but I have questions)
Hello Gabriel, IMHO, using IAX between * servers is a good choice, I dont see any problem in it. Actually I used it for sometime and never encounter any issue, but i had max 5 concurrent connections. regards, Umair bari On 3/7/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi everyone, I just spend the last two hours trying to get two asterisk boxes totransfer calls between eachother using SIP.I dont know why but I *could not* get the calls to authenticate!I think I got everything setup. There was Server A and Server B.I was trying to place a call from ausers registered on Server A to a user regsitered on Server B.I setup the registration info for Server A and even had Server A registeringsuccessfully to Server B.However, whenever I would hand off the calls fromserver A to Server B, it would *always* say it failed to authenticate (passwords did not match).Here was my setup:SERVER A:register = serga:[EMAIL PROTECTED][to_80]username=sergatype=friendsecret=test host=216.152.244.81disallow=allallow=ulawuser=phoneusereqphone=yescanreinvite=yesregseconds=0cancallforward=yesdtmfmode=rfc2833disallow=allallow=ulaw insecure=verytrunk=yesSERVER B:[serga]type=friendusername=sergatrunk=yesnotransfer=yessecret=testcontext=302host=dynamicqualify=yesDIALPLAN ON SERVER A: exten = 302,1,Dial(SIP/to_80/[EMAIL PROTECTED],30,r)It always says authentication failed.However I always noticed it showedthe user as [EMAIL PROTECTED].This is the extension of the phone I am calling from.It seems it is trying to authenticate the actual phone I amcalling from on Server A, and not Server A itself.Was I doing somethingwrong?I tried doing this with IAX and within 5 minutes I had it all working!!I feel it was too easy :-) However, this brings up a big question.IsIAX very reliable for this?I've heard from people that I should not useIAX under any condition because it really is not veryreliable/thourough/consistant...etc.I am trying to start a VOBB company and will obviosly need a reliable setup.I am thinking to have all phonesregister to the servers via SIP and maybe just have all the servers transfercalls between eachother via IAX.Does this sound like a correct setup? - Gabe___--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 handle provider UNREACHABLE in the dialplan?
Dear Michiel,
Would you be kind enough to put more light on RAND stuff. How you do the load balancing.
Regards,
Umair Bari
On 2/4/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On 04:47, Sat 04 Feb 06, Joseph Tanner wrote: This is probably a stupid question, but how do you specify multiple
fallovers?I.e., if provider1 is not reachable/busy, try provider2. If provider2 is down, try provider3.If provider3 is down...etc.I understand how to do it the old way, just keep adding 101 to the
extension.What would you add to a NOANSWER extension though?I guess you could send it to a different context, then you could use another NOANSWER, but I like keeping things short and easy.
[snip] [macro-safedial] ;exten = s,1,Dial(${ARG1},${ARG2},g,${ARG4}) exten = s,1,Dial(${ARG1},${ARG2},${ARG3},${ARG4}) exten = s,2,Goto(s-${DIALSTATUS},1) exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3) exten = s-NOANSWER,2,Hangup exten = s-NOANSWER,3,Verbose(1,Need failover for ${ARG1}) exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Verbose(1,Need failover for ${ARG1}) exten = s-CONGESTION,1,Congestion exten = _s-.,1,Congestion exten = s-,1,CongestionI have this macro too in my
extensions.confLater in the dialplan I use:[outgoing-speakup];dutch telephone nrs.exten = _0X,1,Set(CDR(ACCOUNTCODE)=outgoing-speakup)exten = _0X,2,Set(CALLERID(all)=X)
exten = _0X,3,Macro(safedial,${IAXTRUNK_SPEAKUP01}/31${EXTEN:1},50,Tr)exten = _0X,4,Macro(safedial,${IAXTRUNK_SPEAKUP02}/31${EXTEN:1},50,Tr)exten = _0X,5,Macro(safedial,${ZAPTRUNK}/${EXTEN},50,Tr)
exten = _0X,6,Congestion()Works like a charm.In my production environment I actually load balance callsusing RAND so both IAXTRUNK_SPEAKUP01 and IAXTRUNK_SPEAKUP02get an equal load of calls, but that's not relevant to your
question :)--Michiel van Baakhttp://michiel.vanbaak.info[EMAIL PROTECTED]GnuPG key:
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2DWhy is it drug addicts and computer afficionados are both called users?___--Bandwidth and Colocation provided by
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
___
--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 authorization
Hello Sam, use host=IP_ADDRESS when defining user in sip.conf regards, Umair Bari On 1/26/06, Sam Tam [EMAIL PROTECTED] wrote: Do anyone know how to setup asterisk to authenticate the user through IPrather than username and password? I know most carriers will do that but smaller end user providers will notdo.Sam___--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] 2 PBX linked via internet
www.voip-info.org is one stop ref. for asterisk help. yes its more then possible. You can put asterisk server at your place and make your friend login to your ast box. you can make your friend a sip account use xlite ( windows based softphone ) at your friends place. IMHO you need to buy fxo card to plug your telco or vonage line into ast box. yes thenyour friend can use your vonage or telco line to dialout, and you can make dialplan which actually let your friend chose how to dial and from which line. regards, Umair bari On 12/15/05, Ugo Bellavance [EMAIL PROTECTED] wrote: Hi,I think it is the first time I post on this list,I went throughthe few couple of hundred last messages on the list and couldn't find an answer.I also read the FAQ.I'd need some advice... here is what I plan to do.I've neverinstalled asterisk so I know I need to do some testing.I don't mindbuying equipment, but I don't want to overspend, especially in the test phase.I planned on using [EMAIL PROTECTED] on old boxes that I can get forless than 100$ (P3 500/128 MB) or astlinux on pcengines WRAP machines(or even astlinux on the P3).I just bought a WRAP that is going to eventually replace my netgear home router, but I could use the WRAP fortesting.There is no hurry to replace the netgear.What I want to do is have one asterisk PBX here and one at myfriend's house.We are both with the same ISP, cable modem, with good bandwidth.My friend lives ~ 25 Km from me.- The incoming phone linewould be at my house, but I'd like to have an extension going to myfriend's.I am testing traffic shaping rules with my firewall right now, so this is likely to be in the plan as well.At home I have 2 lines: one regular telco line and one VoIP linefrom Vonage (with a motorola phone adapter).The vonage has unlimitedlong distance USA/Canada (I'm in canada btw).Here are my questions: - Is that possible?Any links to howtos?- Could I connect both of my lines on the pbx?- Can my friend dial through my vonage line to make long-distance calls?Can he actually choose?- I'd like to avoid buying digium cards at least for the testing phase... I know I'll probably need one to use my telco line or the IPline behind the phone adapter, but if I can avoid buying one for myfriend's PBX, it would be great. Can I use (free or inexpensive)softphones with asterisk? Thanks in advance,--Ugo- Please don't send a copy of your reply by e-mail.I read the list.- Please avoid top-posting, long signatures and HTML, and cut theirrelevant parts in your replies. ___--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] I don't want ilbc, i just want G.711
in your sip.cong [general] contexts put disallow=all allow=ulaw allow=alaw and in your sip user, use disallow only ONCE, that is disallow=all allow=ulaw allow=alawhope this helps. regards, Umair bari On 12/15/05, Jason Chan (jasonOfficial) [EMAIL PROTECTED] wrote: Hi there,I am writing to ask about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450 FXO Port, but this gateway just simply doesn't support RFC2833 nor SIP-INFO. The only method I can use isInband DTMF. I know it only support G.711, but I DID disallow others andmake it work only with G.711. But the problem is, although I disallow all other codecs, ilbc still itching me...[extensions.conf][852]username=HKGWserect=blahtype=friendhost=dynamicnat =yescanreinvite=nodisallow=alldisallow=ilbcallow=ulawdtmfmode=inband (P.S. I don't use REINVITE simply because I need the asterisk to be amedia gateway cause the gateway is inside NAT behind the Asterisk)Whenever I try to pass DTMF from phone to Asterisk via that gateway, I got such messages:Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF isnot supported on codec ilbc. Use RFC2833Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?How come!? I DID DISALLOW them, but it keeps bugging me= 192.168.2.3 852 79f9e0-c0a8 00101/1 ulaw No Rx:ACK1 active SIP channel*CLI sip show channel 79 * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 0 Their Codec Capability: 261 Joint Codec Capability: 4 Format ulaw Theoretical Address: 192.168.2.3:5060 Received Address: 192.168.2.3:5060 NAT Support: Always Audio IP: 192.168.2.1 (local) Our Tag: as737358ce Their Tag: 3a53f3e1-bbfcafe6d5c SIP User agent: Username: 852 Peername: 852 Original uri: sip:[EMAIL PROTECTED]:5060 Caller-ID: elite Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5060 DTMF Mode: inband SIP Options: (none)==Previously I installed 1.0.3 in same machine, but I overwrite all files with 1.2.1.. does it cause a trouble?Can anyone figure out what is the problem? ==Thanks very much for your help!Best regards, Jason Chan, Hong KongNo virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005___ --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: [Fwd: Re: [Asterisk-Users] Re: [helpp] Problem in astersik]
Dear Talat,
if you are trying to connect from within your LAN,
put nat=no and then try againregards,
Umair bari
On 12/14/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
-- Forwarded message --From: Talat Ishtiaq
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate: Wed, 14 Dec 2005 15:38:08 +0500
Subject: Re: [Asterisk-Users] Re: [helpp] Problem in astersikHi GuysAfter your guies replies now i have changed the machine .But this time iget little different problemi made following chnages in
sip.conf[901]context=fromsiptype=friendusername=901secret=901callerid=Test2 901host=dynamicnat=yescanreinvite=nodisallow=allallow=ulawdtmfmode=rfc2833
callgroup=3pickupgroup=3qualify=1000;[902];context=fromsip;type=friend;username=902;secret=902;callerid=Test3 902;host=dynamic;nat=yes;canreinvite=no
;disallow=all;allow=ulaw;dtmfmode=info;callgroup=3;pickupgroup=3;qualify=1000in extension.conf[fromsip]exten = s,1,Answer( )exten = _9XX,1,Dial(SIP/${EXTEN},100,tr)
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)exten = h,1,Hangupexten = t,1,Hangupexten = i,1,HangupNowAsterisk 1.0.9, Copyright (C) 1999-2004 Digium.Written by Mark Spencer
[EMAIL PROTECTED]=[ BootingDec 14 15:23:05 WARNING[3478]: chan_oss.c:257sound_thread: Read error on sound device: Resource temporarily
unavailable.Dec 14 15:23:07 WARNING[3478]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled ]
Asterisk Ready.*CLINow from xpro lite software after configuring it for my machine when itry to connect to my machine i am unable to get connection it saysunable to connect contact your network
administratot.Althoug i am thenetwork adminPlz tell me what to doRegardTalatOn Mon, 2005-12-12 at 06:40 -0500, Steven wrote: /var/log/asterisk/full text file may give you a more specific error.
--___--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] Firewall Ports forward
506045691-2 On 12/15/05, Pablo Allietti [EMAIL PROTECTED] wrote: hi all. i have my asterisk with a 192.168.0.1 addresswhich ports i need to forward in my firewall to connect remote xten clients and make calls?thsnk--.-___--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 change the Dial command H option to ## ?
its build into the codes, IMHOfor replacing * with ## you need to hack asterisk source code, I cant think of anyother way. regards, Umair bari On 12/15/05, Obelix [EMAIL PROTECTED] wrote: I want to use '##' to terminate a call instead of the '*' used by the Dialcommand's H option. Is there a way to change the key or use another option to achieve the sameeffect?/ObelixThis message was sent using IMP, the Internet Messaging Program. ___--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] Querry about the modem
look for intel ambient MD3200 chipset in those modems, there were available here in pakistan a year back but now they are vanished from the market. I've been usingmodem with intel ambient md3200 chipsetwith asterisk and they work fine. regards, Umair bari On 11/24/05, Kunhikrishnan, Salil Geethanjaly (STSD) [EMAIL PROTECTED] wrote: Hello Sorry to tell you that I am resending this mail because didn't get a reply for this query. SalilHello I have seen the article in digium site about the answering machine made using a softmodem and the zap library. I am using Fedora Core 2/3 system for doing this project. I was trying to find a PCI modem card with Intel 537 chipset. I couldn't find any model with intel 537 chipset. Can any one please get me some insight into which model I can go for. Available models here in India are, KryptonDlinkIntexAztec... Do any of the above modem have this chipset. Or can I use these models for this purpose.Salil G. K.kpfleming at___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] In France asterisk never detect hang up. Why ?
You can also try busydetect=yes, busycount=4 in zapata.conf. ;; On trunk interfaces (FXS) and EM interfaces (EM, Wink, Feature Group D; etc, it can be useful to perform busy detection either in an effort to; detect hangup or for detecting busies; busydetect=yes;; On trunk interfaces (FXS) it can be useful to attempt to follow the progress; of a call through RINGING, BUSY, and ANSWERING. If turned on, call; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,; so don't count on it being very accurate. Also, it is ONLY configured for; standard U.S. tones;busycount=4 regards, Umair bari On 11/24/05, Faris Raouf [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Hello everybody:-) This are my first line french zapata.conf settings. I have 3 like this, with only rx/tx gain a little bit different levels. Running well. Best Regards, Francois BERGERET, France. usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=6 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=3 busypattern=500,500 signalling = fxs_ks channel = 1 -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] De la part de asterisk user dupont Envoyé : vendredi 18 novembre 2005 13:33 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] In France asterisk never detect hang up. Why ? Hello. I am sorry my english is not good at all. When i have a call from a fxo port of a tdm400p, asterisk waits one minute before detecting that the caller has hang up his phone. I have in my extension conf : answer background(the prompt is 40 second long) dial (on fxs port)confgured for 30 seconds ringing. if the caller hang up at the begining of the background prompt, asterisk waits until he make ring the phone on the dial command for the all 30 secondes before detecting the hang up. Do you know if there is a way to repair that ? here is what i see on asterisk when the caller hang up IMMEDITALY after the test prompt begins : *CLI -- Starting simple switch on 'Zap/4-1' -- Executing Answer(Zap/4-1, ) in new stack -- Executing NoOp(Zap/4-1, 0675458745) in new stack -- Executing Set(Zap/4-1, TIMEOUT(response)=20) in new stack -- Response timeout set to 20 -- Executing BackGround(Zap/4-1, barge) in new stack -- Playing 'test' (language 'fr') -- Executing Dial(Zap/4-1, Zap/2|0675458745|30) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/4-1 -- Attempting native bridge of Zap/4-1 and Zap/2-1 -- Hungup 'Zap/2-1' == Spawn extension (reseau, s, 5) exited non-zero on 'Zap/4-1' -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (reseau, h, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' In my zapata.conf i have : language=fr default=fr relaxdtmf=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 cidsignalling=v23 usecallerid=yes group = 1 context=reseau signalling=fxs_ks callprogress=yes busydetect=yes callerid=asreceived busycount=5 pulse=yes In my zaptel.conf i have : loadzone=fr defaultzone=fr fxoks=1-3 fxsks=4 If anyone can see what is wrong he will really help me. thank you.Your English is better than my French :-) Making the TDM400p detect hangups can be hard. I had it working OK withpre-1.2 versions, but now in 1.2 stable I'm also having some problemsagain. I'll investigate in more details eventually.For now, the only thing I can suggest is that you add: hanguponpolarityswitch=yesin your zapata.confIn the UK, hangups are signaled by a polarity switch, and sincesometimes the UK and Europe do the same thing, I'm hoping this will bethe case for you too. However, even with this option enabled, like I say, I'm having somesmall problems with 1.2 stable. I hope to have time this weekend toinvestigate and see what is going on.Faris.___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cmd dial timeout don't work in asterisk
if you are dialing out from zap channel, you can set the caller id with before dialing out and then you can dial like,
exten = Dial(Zap/2/${EXTEN},15)
and if you are ringing an analog phone connected to fxs port, then you can set the caller id before dialing, then you can dial it like,
exten = Dial(Zap/2,15)
Hope this helps.
On 11/19/05, asterisk user dupont [EMAIL PROTECTED] wrote:
asterisk user dupont wrote: Hello. My dial timeout worked perfectly on the last asterisk but not on the new.
Here is my extension.conf : exten = s,1,Answer() exten = s,2,noop(${CALLERID}) exten = s,3,Set(TIMEOUT(response)=20) exten = s,4,Background(test)
exten = s,5,Dial(Zap/2|${CALLERID},15) exten = s,6,GoTo(personnedispo,s,1) exten = s,106,GoTo(tousoccupe,s,1)Ok problems with the above:1) exten = s,5,Dial(Zap/2|${CALLERID},15) should not contain the | - that is
a separator for parameters, so you are setting a timeout of callerid andoptions of 15.So what do i have to use instead of | ?As i am not in office today, i can not test.. but it seem curios to write :
exten = s,5,Dial(Zap/2${CALLERID},15)no ?I think i must use a separator ?2) If you would like to have +101 bridging you need to use the j option to thedial command now.
I didn't know that.Thank you.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: return Credit Time
Nov 21 06:44:30 WARNING[8266]: Timeout, but no rule 't' in context 'ppp' its not an error its just a warning that you dont have timeout rule in your context, exten = t,1,Hangup OR exten = t,1,Do_SomeThing and it will fix that warning. and about your actual question, use AGI,get customer credit from database, divide that with charges for the number called, it will give you max number of minutes for that destination. Now you have the minutes under customer's credit, you can limit the call.Set max call time limit to that number of minute and you are done. hope this helps. PS: I dont know if above method will work or not, I wrote what I think can solve the issue.Regards, Umair bari On 11/20/05, Abdul Lateef Khan [EMAIL PROTECTED] wrote: Hi,I already install the agiphp from the following steps, i want to besure, is my agiphp installation is correct or not. i copied all following files into/var/lib/asterisk/agi-bin folderphpagi.phpphpagi-asmanager.phpphpagi-fastagi.phpdtmf.php ;For testi crated one extention[ppp]exten = 111,1,agi( dtmf.php)These are all modification which i did for phpagi, Is anotherconfigurations need to be done to work properly?When i am dialing this 111 extentions i am getting the error:Nov 21 06:44:30 WARNING[8266]: Timeout, but no rule 't' in context 'ppp' i will be very thank full if anyone can help me.--Best Regards,Abdul Lateef KhanComputer ProgrammerMobile No. : +974 - 5405022ICQ : 276-994-704YM! : [EMAIL PROTECTED]MSN : [EMAIL PROTECTED]Google Talk : [EMAIL PROTECTED]___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 - Windows Messenger ?
Trywindows messenger 5 http://www.microsoft.com/downloads/details.aspx?FamilyID=16F3A735-FE18-4DF8-9A19-5C6C721CE715displaylang=en Regards, Umair Bari On 11/18/05, Robert Rozman [EMAIL PROTECTED] wrote: Hi,I've found quite some docs on this, but many of them deprecated...I'm curious what is the latest window messenger version that works as registered client to Asterisk... I've tried 4.7, but it registers only if Ileave password empty.Am I missing something or is there any better way to register and useWindows messenger with Asterisk ? Any other sucessful experience with Windows Messenger and Asterisk ?Thanks in advance,regards,Rob.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 - Windows Messenger ?
http://www.microsoft.com/downloads/details.aspx?FamilyID=a8d9eb73-5f8c-4b9a-940f-9157a3b3d774DisplayLang=en sorry about that link, that was a doc. try the link above. regards, Umair On 11/18/05, Robert Rozman [EMAIL PROTECTED] wrote: Hi,I've found quite some docs on this, but many of them deprecated...I'm curious what is the latest window messenger version that works as registered client to Asterisk... I've tried 4.7, but it registers only if Ileave password empty.Am I missing something or is there any better way to register and useWindows messenger with Asterisk ? Any other sucessful experience with Windows Messenger and Asterisk ?Thanks in advance,regards,Rob.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: call status with FXO]
I think that delay in answering is due to caller ID detection.
I have no idea about rest of your question :)
regards,
Umair
On 11/18/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi. I'm a new user of Asterisk. My question is:I want to log outbound calls in a database ( postgres ). Everything is OK
except that asterisk always marks calls to my FXO iface ( Zap/4 ) asanswered as soon as it accepts to dial the specified number and thepurpose of logging calls is to know the actual number of connections made
( those are the ones telco bills me ).I want a method or trick to makeasterisk to behave in regard to a FXO like a modem would do, i.e.reporting line conditions ( BUSY, ANSWER, etc ) as its status. Currently,
the relevant part of my dialplan is:exten = _99.,1,Dial(Zap/4/${EXTEN:2},20)In a related issue, my Zap/4 iface delays answering about 4 secs. When Ifirst installed asterins, answering would be instantaneous, but after some
change ( kernel? compile options? not sure ) it won't answer until after 4or 5 secs. Could it be related to the main problem?Thanks in advance___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can someone explain the 's' extension
No, i really dont think so, here are few lines from extensions.conf.sample. ; Extension names may be numbers, letters, or combinations; thereof. If an extension name is prefixed by a '_'; character, it is interpreted as a pattern rather than a; literal. In patterns, some characters have special meanings: ;; X - any digit from 0-9; Z - any digit from 1-9; N - any digit from 2-9so _X. will only catch digits from 0-9. regards, Umair On 11/14/05, Matt Riddell [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: exten = s is NOT a catchall it's more of a catch nothing i.e. it only catches calls that have no destination info.A catchall would be exten = _.but that would catch extensions that are not numbers (like o, i, t, T, h, etc).A catch all number extensions would be something like exten = _X.Doesn't it catch o, s,h,i,tetc :)--Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html)http://freevoip.gedameurope.com (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk won't listen on another port
try bindport=5062 and bind the IP address too bindaddr=IP_ADDRESS On 9/5/05, Aisling [EMAIL PROTECTED] wrote: Hello, Hope somebody can help me – Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a different port. It is my understanding that I just need to set the port in sip.conf (port=5062) but that doesn't seem to be working. When I type "sip show settings" into the console, I see SIP Port: 5060 in Global Settings. When I run "netstat – tunap" I see: x.x.x.x:5060 LISTEN ser 127.0.0.1 :5060 LISTEN ser 0.0.0.0:2000 LISTEN asterisk . . . 0.0.0.0 :2727 asterisk 0.0.0.0:4520 asterisk 0.0.00:5060 asterisk x.x.x.x:5060 ser 127.0.0.1:5060 ser My config is like follows ;sip.conf [general] context =default port=5062 bindaddr= 0.0.0.0 srvlookup= yes canreinvite= no autocreatepeer= yes [2092] type=friend username=2092 canreinvite= no context=default mailbox=2092 host=dynamic nat= no dtmfmode=info disallow=all allow=ulaw allow=alaw ;extensions.conf ;leave voice messages exten = 2092, 1, Voicemail(u2092) exten = 2092, 2, Hangup ;play voice messages exten = , 1, VoiceMailMain, s2092 ;voicemail.conf 2092 = 2092, 2092, emailaddress At the moment when a user dials to access voicemail, ser forwards to x.x.x.x:5062 and with my current config (port 5062, bindaddr =0.0.0.0) nothing reaches asterisk. However when I change this to (port=5062, bindaddr=x.x.x.x )…the same address as ser, the phones start registering with asterisk even though they're configured to register with port 5060 only! Basically I think Asterisk is still listening on 5060 and I can't change it. I originally thought maybe I had multiple sip.conf's on my machine but when I do "sip reload" in the asterisk console, it says parsing /etc/asterisk/sip.conf, so it's definitely the correct file. Do I need to change the asterisk port somewhere other that sip.conf? Does anyone have other suggestions for what could be making Asterisk behave so oddly? Many thanks, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USING TWO ACCOUNTS WITH BROADVOICE
for taking calls on broadvoice numbers, you need to put insecure=very and need a context in extensions.conf to handle incoming calls on the EXTEN you put in register statement in sip.conf also in sip.conf, put user or friend instead of peer. regards, Umair bari On 9/6/05, Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote: Hi,I have two accounts with broadvoice.Now, I want to be able to distinguish between them. I though that this would be simple by adding /EXTEN at the end of theregister statement. For example:register = num1:[EMAIL PROTECTED]/1000 Unfortunately, this is not working.When I call into my box I hear busy tone.My config looks like this:[EMAIL PROTECTED] asterisk]# cat sip.conf[general]externip=mydomainbindaddr = 0.0.0.0port=5060localnet=192.168.1.0/255.255.255.0disallow=allallow=ulawregister = num1:[EMAIL PROTECTED] register = num2:passsip.broadvoice.comtos=0x18srvlookup=yesnat=neverinsecure=yes[sip.broadvoice.com ]type=peerusername=NUM1fromuser=NUM1authuser=NUM1secret=SECREThost=sip.broadvoice.comcontext=sipfromdomain=sip.broadvoice.com canreinvite=nonat=neverdtmfmode=inband[sip.broadvoice.com.home]type=peerusername=NUM2fromuser=NUM2authuser=NUM2secret=SECREThost=sip.broadvoice.com context=sipfromdomain=sip.broadvoice.comcanreinvite=nonat=neverdtmfmode=inband[broadvoice-incoming]type=peerhost= 147.135.8.128context=from-broadvoicequalify=yescanreinvite=nodisallow=allallow=ulawnat=never[broadvoice-incoming2]type=peerhost=147.135.0.128 context=from-broadvoicequalify=yescanreinvite=nodisallow=allallow=ulawnat=never[broadvoice-incoming3]type=peerhost=147.135.4.128context=from-broadvoice qualify=yescanreinvite=nodisallow=allallow=ulawnat=neverIn addition, I have this config in extensions.[sip]exten = 1000,1,Playback,welcomeThanks in advance. Bart___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Management Interface - Comments requested
Let me know the cost. regards, Umair bari On 8/31/05, Chris A. Icide [EMAIL PROTECTED] wrote: In the next week to two weeks I'll be posting some informationconcerning a system I've been designing.It currently does three layer hosted VoIP pbx services as well as hosted ITSP services (the model isSystem Owner - you, Affiliates - pbx owner/operators or ITSP operators,and end users).The GUI is Zope/python based.The system supports automated clustering, fail-over, and server specialization (you can havea centralized voicemail server if you like, you can centralize sipregistrations if you like, etc.).The Hosted PBX model is in beta testing right now, and we are adding a full billing/provisioning system in September.The bad news, it's not going to be open source.I'll have a web url soon for screen shots and an on-line demo.-ChrisErick Perez wrote: Hi,I want to start managing my asterisk boxes with a centralizedgraphical based interface so I can (due to customers request) givecontrol to customers to add/change extensions to their current PBX intallations such as (not complete list)Add/del/mod extensionssound recordings (ivr or voice attendants)email to fax/ fax to emailvoicemail to emailSIP and ZAP, no IAX needed configure calls routes (server in office A to server in office B, etc)What I want to do is lock them out of the command line (linux) andprovide them with some graphical tool (or some manageable mixture of) that can also help me.This list contains gpl and non gpl providers.So far I have looked at.ACTOSPBX MANAGER from third lane technologiesAMPPBXWARE switchvoxSo far I liked AMP (open source) but switchvox (paid) looks nice too.Comments on AMP integration? I do not want to start adiscussion/flame, I just want some links to AMP modules and see if i can build from different sources a graphical interface that doesend-user pbx functionsBTW call accounting and billing will be nice too.Thanks, ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
Please confirm if PRI span is up on CLI, type pri show span 1 it must be UP before you can dial through it. regards, Umair bari On 8/25/05, root linux [EMAIL PROTECTED] wrote: Yep, I am connecting to some other equipemnt...its a Clarent gateway equipped with a National Microsystems (Quad Port) --- El Flynn [EMAIL PROTECTED] wrote: Hi there, Are you getting the E1 span in from Telekom, or are you connecting to some other equipment? root linux wrote: My zaptel.conf config: - # Below setting is for E1 span=1,1,0,cas,hdb3 bchan=1-15 bchan=17-31 dchan=16 loadzone = us defaultzone=us My zapata.conf config: - # Below setting is for E1 switchtype = national signalling = pri_cpe group = 1 channel = 1-15 channel = 17-31 My extension.conf config: - [default] exten = 181,1,Dial(Zap/1/181) When I perform a dailing from my SIP Phone, I got the error message as below: - -- Executing Dial(SIP/118301-6f4e, Zap/1/181) in new stack Aug 22 11:03:02 NOTICE[9023]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time I am beginner...How to solve this? Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to change Port for SIP users
Dear All, I want to change port for SIP users in asterisk realtime, My ISP has blocked 5060-5062 ports ports, i tried making asterisk listen to 5069 and it worked but when i assigned ports to sip users in sip_buddies, it didnt work. Clients are not loggin in. Any solutions? Thanks in Advance, Regards, Umair bari ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse DTMF Problems Anyone?
Michael, try relaxdtmf=yes in your iax.conf, or if you are using sip, then in sip.conf regards, Umair bari Michael Stearne wrote: On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: That's entirely possible. Had something similar with livevoip.com (with the answered iax call issue). You should be able to determine whether its a voicepulse issue by either doing a iax debug (look for the dtmf digits), or, using ethereal to trace the packets. Both methods should show the pressed dtmf digits as values passed in the iax frame. If you don't see those, then its likely voicepulse is passing the dtmf tones as audio (try different codec). Thanks! I'll try that. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Umair Bari Tech Support Dept. Super Technologies Inc. http://www.supertec.com Voice : 1-408-884-1966 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk eating up 99.8% cpu
Dear All, I am using Asterisk CVS-HEAD-05/05/05-15:46:09, and found it using up 99% of cpu 6-8 times in a day, even when its doing nothing or even if it is, its not supposed to eat all the cpu. whenever this happens, I need to stop it by issueing stop now 15-20 times on CLI or kill the process, then start over. Please suggest what to do. regards, Umair bari ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rings - How to set number
exten = 1234,1,Dial(SIP/1234,Number_of_Sec_for_Ringing,tr) Tim P wrote: Maybe this marks me as a real newb but where do I set the number of rings that a phone has before it sends it to voicemail? Also for some odd reason when I ring an extension attached to my sipura 2100 ATA it takes it about 12 seconds to start ringing after I dial it (sits there with dead air on the calling phone). Any idea on these, am I missing some simple configuration switch for either? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP - PSTN call)
try putting
exten = _0.,4,Hangup
like
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
exten = _0.,4,Hangup
regards,
Umair bari
Tomasz Chmielewski wrote:
I'm trying to learn Asterisk.
So far I'm using kphone and an ISDN line (with Eicon DIVA 2.01 PCI card).
I have created that extension following The Asterisk Handbook (page 36):
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
So whenever I call 055 from kphone, Asterisk connects me to an
internal 55 number, and I can talk to myself (wohoo!) when I pick up
the phone.
However, when I call 055 from kphone, and *don't* pick up the phone on
the other side, and then disconnect kphone (or even quit it), asterisk
keeps ringing 55.
I'd like to add, that Asterisk detects kphone disconnecting when the
phone is already established.
Any clue?
Tomek
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended Linux Dist. for Asterisk
Using RH 9 with * Regards, Umair Bari David Choo wrote: We used gentoo internally. I also have * running on CentOS, RHEL. Best Regards, == David Choo Systems Engineer Business Technology Division "Engineered for Changing Businesses" Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Michael George [EMAIL PROTECTED] a.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 21/04/2005 10:31 Re: [Asterisk-Users] Recommended PMLinux Dist. for Asterisk Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Wed, Apr 20, 2005 at 10:26:33PM -0500, Paul Shiflet wrote: I'm trying to find out what flavor of Linux people are choosing for their asterisk boxes. I have been using RH, but i'd like to try some different ones. It seems that RH is the common denominator in this rash of line noise problems. So some suggestions for what dist to use would be great. We use gentoo. Many people would not go that route, but we use that on our servers because when we are ready to update it, we can do so with less pain than with RHL/Fedora and SuSE, etc. The updates of the latter usually go okay, but there comes the time when we need to change major releases and that should be done with a clean reinstall. Now, with * you don't really need to do any changing as it will just sit there and work for the most part. However, since we have gentoo in many of our systems, we just stick with that. The ports in gentoo stay pretty current and it's worked fine for us. YMMV, and as I said above, gentoo is probably not the route for many who have little linux experience. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
