Re: [asterisk-users] Call Center SoftPhone with Auto Answer
X-Lite do what you need... On 8/6/07, Joao Pereira [EMAIL PROTECTED] wrote: Hello I need a Softphone with auto answer where users can't turn it off. Does someone knows a softphone where users can't turn the auto answer off? Or is there any way Asterisk could force the clients to answer the phone? Thanks Regards Joao Pereira ___ --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 1.4.1 app_addon_sql_mysql
Do you have MySQL installed in your machine??? On 6/21/07, Khaled Chehab [EMAIL PROTECTED] wrote: No one faced a problem like this !! From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql …. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Transfer caller direct to voicemail
Hello Drew;
Assuming your extensions is 105 let's see the dialplan:
exten = 105,1,Dial(SIP/105,30,Tt)
exten = 105,n,Hangup
exten = *XXX,1,Answer
exten = *XXX,n,VoiceMail(${EXTEN:[EMAIL PROTECTED])
exten = *XXX,n,Hangup
I think this should work for what you want.
Regards;
Leonardo Kamache
Rio de Janeiro - Brasil
On 6/12/07, Drew Gibson [EMAIL PROTECTED] wrote:
Hi,
Our operator frequently gets requests to transfer a call directly to
voicemail in order for the caller to leave a message without disturbing
the callee. Basicly, I'm looking for a blindxfer to vm.
My first thought was to prepend a digit (eg 7) to the extension but this
does not fit well with our dialplan.
According to an article on voip-info.org [EMAIL PROTECTED] appears to
implement this as #*XXX. I assume they are using an application map in
features.conf but I cannot see a way to pass the required extension to
the VoiceMail() application.
Can this be done in features.conf?
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
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Re: [asterisk-users] Softphone behind NAT issues
In [general] section: externip=your_extern_ip_address localnet=your_local_net/bits i.e. 192.168.0.0/24 Try this... On 6/12/07, Rob Schall [EMAIL PROTECTED] wrote: We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching the traffic and noticed that there doesn't appear to be any rtp traffic going back to asterisk (this is where we think the problem might be). The firewalls on both sides have ports 5060, 1-2 and 3478 (STUN) open. Out conf files are: -- [sip.conf] [general] context=incoming; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls allow = all [1000] nat=yes type=friend secret=Polycom context=internal host=dynamic canreinvite=no [EMAIL PROTECTED] callerid=TESTUSER1 1000 - [extensions.conf] exten = 1000,1,Macro(stdexten,[EMAIL PROTECTED],SIP/1000) [rtp.conf] [general] rtpstart=12000 rtpend=12005 dtmftimeout=3000 What are we missing? ___ --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] Voip-info.org
Yes from Brazil... On 6/6/07, Ed Nuñez [EMAIL PROTECTED] wrote: Is anyone else having trouble going into voip-info.org today? ___ --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] spa 3102 incoming call
Hi Damiano! Take a look at this link: http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5159lid=6862769263B11 Best regards; Leonardo Kamache On 6/5/07, damiano bertuna [EMAIL PROTECTED] wrote: Hi to everybody, I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port). This is my problem: the incoming call doesn't arrive to asterisk. In the spa web page i configured this dialplane: (:[EMAIL PROTECTED]:5060) where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip and 5060 is the asterisk sip port. [line01] username = usersipura fromuser = usersipura secret = pwdsipura host = 192.168.1.222 fromdomain = 192.168.1.222 port = 5061 type = friend dtmfmode = rfc2833 context = call_in insecure = very Why? is the dialplane wrong? help me, please. Damiano. ___ --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] Call Pick Up
Two words for you... parking lot. Try to transfer your call to extension 700 and see what hapens... On 4/27/07, Jim Duda [EMAIL PROTECTED] wrote: I use Asterisk in my house. Each phone is a different extension. I really like the ability to have multiple simultaneous calls in the house. However, I do miss being able to be able to pick up a phone in a different room. Currently, I have to either transfer the call or transfer the call to a conference extension to move around the house. While a connection in progress on one extension, I would like to go to any other phone, dial some extension number, in order to ether pick up the call or join in an automatic conference. In other words, make it work like the old ma bell phone (when I want it to :-) ) Is this possible in Asterisk? Thanks, Jim ___ --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] Transer calls hitting #
Try to configure your PAP2 DTMF send mode to INFO. On 4/21/07, Doug Lytle [EMAIL PROTECTED] wrote: Poul Moller wrote: Are there any special ATA audio setting I should apply? That I don't know, I've never setup an ATA before. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Why duoble digits must be so fast to activate features?
Hi Mauro; Try to add featuredigittimeout = 1500 at features.conf in the [global] section. On 4/20/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 20 Apr 2007, Mauro Zanin wrote: Hi everybody, I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm using Trixbox..). I must be as fast as a flash to press *2 and do an attended transfer. If I wait only a tenth of a second nothing happens. I think it is an issue. I have seen the source code and found nothing bad. Is this a known issue? Change it in features.conf. Gordon ___ --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] internal sounds of asterisk / freePBX
Did you have any E1/T1 cards in your server? On 4/18/07, shadowym [EMAIL PROTECTED] wrote: CallWeaver is the new name for OpenPBX -Original Message- From: Carlos Jerónimo [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX i use xlite and kphone in a diferent pc's. i can phone well. the problem is internal asterisk sounds. I think i not use Call Weaver, what is call weaver, i search at google but i'm was confused. i hope more help's. thanks 2007/4/17, Andrew Joakimsen [EMAIL PROTECTED]: If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --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 --
Re: [asterisk-users] Newbie Question
Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box. Best Regards; Leonardo Kamache On 3/8/07, Dovid B [EMAIL PROTECTED] wrote: If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --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] Good Book on Asterisk
Hi Robert!Did you try http://www.voip-info.org ?Regards;Leonardo KamacheOn 9/28/06, Norbert Zawodsky [EMAIL PROTECTED] wrote: Michel Vaillancourt wrote: Norbert Zawodsky wrote: Hi everybody! I have some Linux experience but I'm completely new to asterisk. I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk (1.2.12) preinstalled and some basic configuration (Wiht a few extensions). Now I want to implement something more, fox example voicemail (storing voicemail data in an extern mysql DB) and so on. And since I don't want to waste your time with stupid questions ... can someone of you recommend a really good book on Asterisk? (To buy or for download) ... or another online source of information which would be helpful for someone like me? I searched Amazon with Asterisk and got 21 hits.. Thanks Norbert Hi, Norbert ... The O'Reily Book for Asterisk: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Enjoy!Thanks for the book. I read it last night (or nearly all of it) and now i think i understand a bit more.But now the next question:Where can I find the documentation of the applications and functions Ican use in the dialplan? (For example how to use the mysql add-on, ...) ThanksNorbert___--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] Where can i get a telephone number of Brasilia or Rio de Janeiro in Brazil
Hello Han! I'm from Rio de Janeiro and I'm using Tmais (www.tmais.com.br). I'm very happy with the service and they accept credit card payment. Regards; Leonardo Kamache On 8/17/06, Han van Hulst [EMAIL PROTECTED] wrote: Who can help me out i am looking for a Brazilaan telephone number in the City's Brasilia and Rio de Janeiro. I am looking for a local provider with creditcard payment. Thank Han ___ --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] Page()
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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