[Asterisk-Users] No dial tone
Hello all, My problem is that when I call from an extension to another, I ear the dial tones, but when I make a call using the Zap or Capi channels I do not ear the dial tones. Why this could be happen? Any clue will be appreciated. Thanks. Ismael gil. ___ 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] Zaptel config
Hello all, I am having a lot of problems with zaptel channels, I have got an TDM02B, and I don't know how setup /etc/zaptel.con and /etc/asterisk/zapata.conf for use it on asterisk. Some one could help me with this configuración? My problem is about the type of signalling Thanks, Regards. Ismael Gil. ___ 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] TDM box Hardware
Hello all, Recently I bought a TDM02B digium card to conect to the PSTN. We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try to start asterisk, the box hangs. Someone have the same card running with asterisk in a similar machine? Could you tell me your box hardware details? Thanks for your time, Ismael Gil. ___ 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] Zaptel problems
Hello all, I recently install a TDM04B with only 2 FXS modules. I just install Asterisk 1.0.3, zaptel 1.0.3, and libpri 1.0.3, all of these stable packages. I configured /etc/zaptel.conf like the following loadzone = us #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no defaultzone=us fxsks=1-2 And the /etc/asterisk/zapata.conf like this. signalling=fxs_ks context=incoming channel = 1 signalling=fxs_ks context=incoming channel = 2 One of this for each channel. But when I start asterisk with asterisk -vvvcf, the machine where asterisk is instaled hangs up. Asterisk froozen the machine. Anybody knows how to solve the problem? Any clue will be appreciates. Regards, Ismael Gil. ___ 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] More Zaptel problems
After this config, I just load the modules, doing that, modprobe wcfxs modprobe zaptel Then I make ztcfg, and I get that Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. But when I try to start asterisk, I get a box crash just after parsing musiconhold. Any clue? Thanks Ismael Gil. ismaelg wrote: Hello all, I recently install a TDM04B with only 2 FXS modules. I just install Asterisk 1.0.3, zaptel 1.0.3, and libpri 1.0.3, all of these stable packages. I configured /etc/zaptel.conf like the following loadzone = us #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no defaultzone=us fxsks=1-2 And the /etc/asterisk/zapata.conf like this. signalling=fxs_ks context=incoming channel = 1 signalling=fxs_ks context=incoming channel = 2 One of this for each channel. But when I start asterisk with asterisk -vvvcf, the machine where asterisk is instaled hangs up. Asterisk froozen the machine. Anybody knows how to solve the problem? Any clue will be appreciates. Regards, Ismael Gil. ___ 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] More Zaptel problems
It crashes, here the last messages asterisk wrote in stdout before hangs. [app_hasnewvoicemail.so] = (Indicator for whether a voice mailbox has messages in a given folder.) == Registered application 'HasVoicemail' == Registered application 'HasNewVoicemail' [format_wav_gsm.so] = (Microsoft WAV format (Proprietary GSM)) == Registered file format wav49, extension(s) WAV|wav49 [app_url.so] = (Send URL Applications) == Registered application 'SendURL' [chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver) [app_test.so] = (Interface Test Application) == Registered application 'TestClient' == Registered application 'TestServer' [skipping chan_mgcp.so] [app_eval.so] = (Reevaluates strings) == Registered application 'Eval' [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Thanks. Ismael Gil. Alessio Focardi wrote: i Channel 01: FXS Kewlstart (Default) (Slaves: 01) i Channel 02: FXS Kewlstart (Default) (Slaves: 02) i 2 channels configured. i But when I try to start asterisk, I get a box crash just after parsing i musiconhold. i Any clue? * hangs or crashes ? And if it crashes: have you got any message on screen ? ___ 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] Reading mysql sip friends
Hello, I am trying to setup an asterisk to store users datails in a mysql database. Explained here http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers All works well, I create a user in the database, but asterisk seems that can't read the data, I create a mysql user with enough permissions. Is needed do anything more to obtain asterisk read the sip config from database? What I am doing wrong? Thanks Ismael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mysql configuration interface
Hello, I trying to configure asterisk to store sip and iax2 user in a mysql database. All goes well, but my problem is when i try to add a new user (sip or iax). I have look for an aplication with a web interface that lets us manage the user account in asterisk without success . How could I manage the users without making sql query throw the mysqlclient? Do you know any application that provide this functionality? Is needed for use the mysql support for asterisk, to make our own web appication to manage the users? Thanks. Regards from Spain. Ismael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A waning console error
Hello, I am getting this kind of Warning in the Asterisk console, but i don't know why. WARNING[8200]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Could you give some clue to solve this problem? Thanks in advice. Ismael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding calls
Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on the line until the extension timeout expired. What I want is when I dial am extension currently Busy (Talking with someone), asterisk inmediately forwards my call to an extension I previosly defined. Someone could help me? Any clue will be appreciated. Regards from Spain. Ismael Gil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] edirecting calls with Asterisk
Hello, I am trying a couple of days before to set up asterisk to redirects an incoming call if the extension dialed is busy without success. I just try to use 'Gotoif' command, with bad luck, it can't do what i want. Anybody could helpme? ani clue will be appreciated. Regards. Ismael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 DNS issue
Hello all, I just configure Bind 9 in our LAN to resolve the Asterisk name sip.bussines.com for our phones. I want that when a local extensión calls to another local extension, the phone shows Extension@DNS name instead of Extension@ip address like now happens. In all my phones I configure the sip server like sip.bussines.com (dns name), but I don't know how to get it. Someone could give me some hint? any clue will be appreciated. Thanks in advice. Ismael Gil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I can't solve my problems with the IVR
Hello all, I'm still having problems with the IVR options. When I press on my mobile phone one of the digits related in the IVR options, press 1 for .,press 2 for.., press 3 for.. After I press the one, the second or the tirth key on my mobile phone, I can't hear nothing more, I can't hear the following menu. I just search info about dtmf but i can't find information witch help my to solve my problem. Any clue will be appreciated. Here is my channel definition in Zapata.conf signalling=fxs_ks callwaiting=yes language=en context=incoming callerid=asreceibed relaxdtmf=yes channel =1 And here a user defined in SIP.conf (All users I have have the same config) [pepe] type=friend ;secret=lele host=dynamic ;dtmfmode=inband; Choices are inband, rfc2833, or info dtmfmode=info defaultip=xxx.xxx.xxx.xxx mailbox=122 ; Mailbox for message waiting indicator ;restrictcid=yes; To have the callerid restriced - sent as ANI pickupgroup=1 callgroup=1 username=pepel ; usually matches the section title fromuser=22 ; overrides the callerid, e.g. required by FWD callerid=pepe Thank you in advice Ismael Gil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR option problem
Hello all, I'm trying setting up an IVR on a Asterisk Soho PBX. My problem is when I dial the IVR extensin from an Asterisk internal extension all goes well, but when I dial the external number of the IVR, e.g. 119235656, the PSTN number of my asterisk, I get the same IVR menu but when I press on my phone the 1, to select the fist IVR option, or 2, to select the second one, (the IVR has only two options), I can't hear anything more on my phone. My phone gets silent. And I lost the rest of IVR locution. Regards. Ismael. Gil. Steven Critchfield wrote: On Mon, 2004-10-18 at 13:35 +0200, [EMAIL PROTECTED] wrote: Hi.Is possible to caprure calls with asterisk?I have a calling from onde device to another. While its ringing Idwish to capture the calling from another device which has permissionsto make it. is it possible? Check out pickup groups. BTW Digest users should be strongly urged to convert to normal messagesas your less likely to make stupid mistakes with regards to responses.Like when you forget to TRIM irrelavent sections of the message, itdoesn't force us all to rereceive large numbers of messages. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A question with voice Menu
Hello, I'm having the following problem in my asterisk config. I have a little voice menu, with two options, The welcome message looks like that, 1- press 1, to dial an extension 2- press 2, to speak with an operator. If I press 1, I get the following message Dial the extensión number you want to talk to... But if I wait a moment after this message I get this message again 1- press 1, to dial an extension 2- press 2, to speak with an operator. Asterisk repeat the welcome message again, and this isn't what we want. How could I solve this? Thanks Ismael. (I just Paste the config) [incoming] exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto(contexto_extensiones,s,1) exten = 2,1,Goto(contexto_operadora,s,1) [contexto_operadora] exten = s,1,Background(itranser/trans_operadora) exten = s,2,Dial(SIP/aurelio,100,Ttr) [contexto_extensiones] include = default exten = s,1,Background(itranser/msg_pasar_ext) exten = s,2,Wait,Ttr,200 The dafault context is where I defined all my phone extensions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changing the default language
Hello all, I am tring to change the default language in Asterisk, exactly for the Voicemail messages. I trying with the option Language=fr in the voicemail.conf global section, without success. I trying with the Setlanguage(fr) in the extensions.conf global section, but without success too. How could I change the default Languaje for Voicemail? I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have a letter and diggits directory too. Any clue will be appreciated. Regards. Ismael Gil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with voice menu
Hello all, I having a lot of troubles to configure a simple voice menu. In extensions.conf I have the following. [incoming] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto,contexto_extensiones exten = 2,1,Goto,contexto_operadora The context refered by the menu. (each context play me a diferent message only ) [contexto_operadora] exten = 2,2,Background(itranser/trans_operadora) exten = 2,3,Dial(SIP/ismael,s,1) [contexto_extensiones] exten = 1,1,Background(itranser/msg_pasar_ext) My problem, is when I touch the key 1 in my phone, after the msg_bienvenida, asterisk do not pass me to the correct context [contexto_extensiones]. Asterisk do not pass me to any context, asterisk do nothing when I press the 1 key on my phone. Have I missed something in my extensions.conf? or in sip.conf? Thanks Regards from Madrid. Ismael Gil.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Problems with voice menu
Thank you Christopher,
Imade the changes you told me, but, when I try to make an incoming call,
in the Asterisk console, I get
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569/9'
-- Executing Dial("SIP/aurelio-92fe", "IAX2/501050:[EMAIL PROTECTED]/501050|60|r")
in new stack
-- Called 501050:[EMAIL PROTECTED]/501050
-- Call accepted by 65.39.205.121 (format ULAW)
-- Format for call is ULAW
-- Accepting AUTHENTICATED call from 65.39.205.121, requested format
= 4, actual format = 4
-- Executing Goto("IAX2/[EMAIL PROTECTED]:4569/14", "incoming|s|1")
in new stack
-- Goto (incoming,s,1)
-- Executing Wait("IAX2/[EMAIL PROTECTED]:4569/14", "1") in new stack
-- Executing Answer("IAX2/[EMAIL PROTECTED]:4569/14", "") in new stack
-- Executing DigitTimeout("IAX2/[EMAIL PROTECTED]:4569/14", "10")
in new stack
-- Set Digit Timeout to 10
-- Executing ResponseTimeout("IAX2/[EMAIL PROTECTED]:4569/14", "20")
in new stack
-- Set Response Timeout to 20
-- Executing BackGround("IAX2/[EMAIL PROTECTED]:4569/14", "itranser/msg_bienvenida")
in new stack
-- Playing 'itranser/msg_bienvenida' (language 'en')
-- IAX2/65.39.205.121:4569/13 answered SIP/aurelio-92fe
-- Channel 'IAX2/[EMAIL PROTECTED]:4569/14'
unable to transfer
-- Hungup 'IAX2/65.39.205.121:4569/13'
Why I get an "Unable to transfer" error on this channel?
How could I solve this problem?
Any clue will be wellcome
Thanks a lot.
Ismael Gil.
Christopher Lee wrote:
I having a lot of troubles to configure a simple voice menu.In extensions.conf I have the following.[incoming]exten = s,1,Wait(1)exten = s,2,Answerexten = s,3,DigitTimeout,10exten = s,4,ResponseTimeout,20exten = s,5,Background(itranser/msg_bienvenida)exten = 1,1,Goto(contexto_extensiones,s,1)exten = 2,1,Goto(contexto_operadora,s,1)The context refered by the menu. (each context play me a diferent message only )[contexto_operadora]exten = s,1,Background(itranser/trans_operadora)exten = s,2,Dial(SIP/ismael,s,1)[contexto_extensiones]exten = s,1,Background(itranser/msg_pasar_ext)
I've made the corrections to your context's above... Note in particularthe Goto command and then using the 's' (start) extension in eachextension line, also adjusted the priority numbers. For more info on Gotohttp://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20GotoGive that a try and see how you go.Regards,Chris Lee___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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