Re: [Asterisk-Users] CallerID for BSNL (India) phones
Hi raj, Perhaps both of us are going through same tunnel... some days back there was a thread , sorry that I am repeating it here * Hi! Looking at the CID code for ss_thread() in chan_zap.c (CVS HEAD as of yesterday), it appears pretty clear that there is no way it will work with the Dutch POTS network - the caller-id is sent as DTMF *after* the first ring. I thought this had worked in some version, but has it been broken again in the latest ones? Julf * As india also uses dtmf detection...as dutch POTS so this may be the reason. Even if it is not the reason then developers pls guide to achieve the goal. /Guri On 10/11/05, Rajkumar S [EMAIL PROTECTED] wrote: Hi, What must be done to enable callerid and call progress monitoring (disconnect notification) for Zap lines connected to BSNL phones in India. I am willing to get documentation, test or write the necessary code to get it working. I have gone through the indications.conf, will that be sufficient? Any one to help me get there? raj ___ --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
Re: [Asterisk-Users] Cannot dial SIP via asterisk
Hi Obelix, You can check that network is working fine. The ipaddresses in sip.conf are correct. /Gurmi On 10/9/05, Obelix [EMAIL PROTECTED] wrote: I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Destroying call '[EMAIL PROTECTED]' O This message was sent using IMP, the Internet Messaging Program. ___ --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
Re: [Asterisk-Users] [help!] asterisk 1.2 beta
Hi ryan, Have u used safe_asterisk which runs asterisk as a dameon. try it putting in yur startup script. regards /Gurmi On 10/6/05, ryan nalupa [EMAIL PROTECTED] wrote: hello! i've tried to use asterisk 1.2 beta version and all installed fine except that when i config asterisk to run at bootup it gives me an error which says 'execvp:Permission Denied!' then the loading fails so i stil has to start asterisk manually when i've logged in as root which is too much work, i want asterisk to initialize immediately on bootup. can someone help me in figuring that out? thanks in advance! :) __ 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 enter digits using sjphone
Hi all, A small question relating sjphone Here it is I am connecting from pc to Asterisk using Sjphone. I can make outgoing calls according to dial plan setup, but I am not able to enter options asked during the call like enetering passwords for voicemail. SJ phone initiates just another call for it like [EMAIL PROTECTED] I tried with many other options but definetley lacking the key :( Also if any body can tell me about other good softphones for linux I know abt kphone, Iaxcomm, twinkle, linphone, sjphone regards /Gurmi ___ --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] IAX2 calls dropped (Max retries)
Hi all, An IAX2 question... Working on two Asterisk machines using beta1. Lets say A and B A = B (using IAX2 drops call) It prints warning message of Max retries exceeded . B = A (works fine as it should) I searched the archives and got similar question which is unanswered. Surely it cannot be network problem because it is going one way. Can it be with port 4569 on outgoing channels of one machine. Any help any idea:) /Gurmi ___ --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] sip phones on x86_64
me too looking for softphone...not able to enable kphone Can anyone please highlight more on it. ThX /Gurmi On 10/4/05, Wayne Gemmell [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 00:42, Rajesh kumar wrote: I am using Kphone which works great for my purposes! You can look at twinklephone as well at http://www.twinklephone.com/ Hi, thanks all for the info, kphone does really wierd stuff and I can't get twinkle to compile. I'm looking into that gnomeeting CVS idea. -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] analog phone connects to zaptel fxoks is beeping
Check if there is any priority conflict between extensions in you outgoing context in extensions.conf:) Regards /Gurmi On 10/3/05, Michael Jia [EMAIL PROTECTED] wrote: Hi, I have a analog phone connect to a WCTDM card. It used to work fine. Now recently, after several conf change and power restart, it stops working. Whenever I pickup the phone, instead of hearing the dial tone, I hear a busing beeping tone, like a machine gun is firing. :) However, from asterisk console, I do see a a OffHook/OnHook message, but whatever I dial in the phone keypad seems not recognized by asterisk, and it doesn't print out any messages. What could cause the problem? Any clues? Highly appreciate your help. Michael ___ --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
Re: [Asterisk-Users] What does the error stale nonce' mean?
I'm receiving the following error over and over, adnauseam: Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID sip:[EMAIL PROTECTED]' In message itself no where it is written ERROR But thanks to Stewart and Olle for giving in depth information. /Gurmi ___ --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] Do Sifira use Asterisk?
Hi, I am looking for what sifira use to provide its services like Callrecorder, Family voicemail..etc Does it uses Asterisk, if yes, for what specific services? Thanks Gurminder ___ --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] Re: MusicOnHold not working
Hi, Thanks all for help... I was perhaps using old version of mpg123 and with beta1 and mpg123-0.59r it working smoothly. Gurminder On 9/15/05, Gurminder Arora [EMAIL PROTECTED] wrote: Hi On my FC3 box with asterisk 1.0.9MusicOnHold is not working. It starts and stops immediately... An unknow option mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any ideas any direction... Thanks Gurminder Console output ***Snip*** -- Executing MusicOnHold(Zap/1-1, default) in new stack -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 Unknown option: --mono *** My musiconhold.conf is ***Snip [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 application=/usr/local/bin/mpg123 Snip** ___ --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] Hangup after voicemail not detected
Hi, On my FC3 box I am having *v1.0.9. The problem is that when a user calls through POTS line and leaves a message in voicemail, the channel doesn't detect the remote hangup. After 10 seconds of remote hangup it plays messages like vm-thankyou, vm-review etc as if user is still online. After that it hangs up displaying a warning message WARNING[25623]: file.c:568 ast_readaudio_callback: Failed to write frame Can it problem related to service provider...?? Had anyone solved this kind of problem earlier or if have any idea...Please help! My extensions.conf incoming context is [incoming] exten = s,1,Answer exten = s,2,Dial(Zap/1,5,tr) exten = s,3,Voicemail([EMAIL PROTECTED]) exten = s,4,Hangup * Gurminder ___ --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] E1 configuration problem
Hi, First of all make sure you have inserted zaptel and wxfx* modules and Then execute ztcfg -vvv It will show you the channels configured and see its green signal now. :) Regards Gurminder On 9/19/05, manish kumar [EMAIL PROTECTED] wrote: I am trying to configure E1 card (Digium) but not able to do that. The green light doesn't come up when it starts. What can be the problem. I have also changed the jumper settings of the card from T1 to E1 but still no relief. Thanks in advance Manish ___ --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
Re: [Asterisk-Users] MusicOnHold not working
Hi, I am using asterisk 1.0.9 on FC3 box. mpg123 is not working it starts and stops immediately.. printing - -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 Unknown option: --mono There is no such option defined in my musiconhold.conf file After searching through code: I got --mono option defined in function spawn_mp3(struct mohclass *class) in file res_musiconhold.c After commenting that line message stopped but mpg123 still didn't responded nd I am trying to find. I am wondering if mpg123 doesnot support --mono option They why it is forced in asterisk Code? Thanks Gurminder On 9/15/05, Alex Kobalto [EMAIL PROTECTED] wrote: I have the same problem with several softphones (Xlite), but there's one, Firefly I think, that worked. I found it strange, but not a real problem for me. I have the same asterisk server version, wich came with the last [EMAIL PROTECTED] distribution. On 9/15/05, Sherwood McGowan [EMAIL PROTECTED] wrote: It's because mpg123 is being passed an option --mono. Looks to me (a cursory guess) that your asterisk installation is trying to force mono sound, and mpg123 doesn't like it. --Original Message- -From: [EMAIL PROTECTED] -[mailto: [EMAIL PROTECTED] On Behalf Of -Gurminder Arora -Sent: Thursday, September 15, 2005 8:30 AM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] MusicOnHold not working - -Hi - On my FC3 box with asterisk 1.0.9MusicOnHold is not working. -It starts and stops immediately... -An unknow option mono comes...from where it is originating.?? -As there is nothing written in .conf file. -Console output is below: - - -I am using mpg123 version 0.59r. -Although I am able to play music with mpg123 but why it is on -No-cooperation movement against asterisk ? - -Need help..any ideas any direction... - -Thanks -Gurminder - - -Console output -***Snip*** --- Executing MusicOnHold(Zap/1-1, default) in new stack --- Started music on hold, class 'default', on channel 'Zap/1-1' --- Stopped music on hold on Zap/1-1 -Unknown option: --mono -*** - - - -My musiconhold.conf is -***Snip -[default] -mode=mp3 -directory=/var/lib/asterisk/mohmp3 -application=/usr/local/bin/mpg123 -Snip** -___ ---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 ___ --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] MusicOnHold not working
Hi On my FC3 box with asterisk 1.0.9MusicOnHold is not working. It starts and stops immediately... An unknow option mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any ideas any direction... Thanks Gurminder Console output ***Snip*** -- Executing MusicOnHold(Zap/1-1, default) in new stack -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 Unknown option: --mono *** My musiconhold.conf is ***Snip [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 application=/usr/local/bin/mpg123 Snip** ___ --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] Doesn't finishes callerid spill
Hi,
I am a beginner in asterisk. Implementing it in my dept in India
using TDM400b card with asterisk, zaptel, libpri version latest of CVS
HEAD
Callerid on my system is coming tough.
Asterisk doesnot finishes the callerid spill and Cancells it.
After going through code in Callerid.c and chan_zap.c I found that my
line is providing caller id of length 8867.
Flow enters in zt_call and generates callerid id of length 8867 from
callerid generate in callerid.c
*snip** zt_call** chan_zap.c**
if (p-cidspill)
p-cidlen = ast_callerid_generate(p-cidspill, ast-cid.cid_name,
ast-cid.cid_num, AST_LAW(p));
p-cidpos = 0;
send_callerid(p);
//Flow enters in send callerid in a while loop which checks
cidposcidlen; Initial cidpos=0 and cidlen =8867
***snip** send_callerid*chan_zap.c
//
while(p-cidpos p-cidlen) {
if(!p-cidpos)
{
res = write(p-subs[SUB_REAL].zfd, p-cidspill + p-cidpos,
p-cidlen - p-cidpos);
//res here comes out to be 160
}
if (res 0) {
if (errno == EAGAIN)
return 0;
else {
ast_log(LOG_WARNING, write failed: %s\n, strerror(errno));
return -1;
}
}
if (!res)
return 0;
// res increments pos by 160
p-cidpos += res;
}
*
The strange thing happens here when loop is executed 35-37 times
cidpos is inreased to near about 5700 8867 and suddenly control gets
in zt_handle_event function in a switch case statement and cancells
the callerid spill and continues.
***snip***zt_handle_event***chan_zap.c*
case ZT_EVENT_RINGEROFF:
if (p-inalarm) break;
if (p-radio) break;
ast-rings++;
if ((ast-rings p-cidrings) (p-cidspill)) {
ast_log(LOG_WARNING, Didn't finish Caller-ID spill. Cancelling.\n);
free(p-cidspill);
p-cidspill = NULL;
p-callwaitcas = 0;
}
p-subs[index].f.frametype = AST_FRAME_CONTROL;
p-subs[index].f.subclass = AST_CONTROL_RINGING;
break;
***
I am seaching Why loop exits before reaching limit of 8867 or what
makes zt_handle_event to control the flow.
Please help me with any idea you have. Also tell if I am on wrong path
for right problem
PS: I have tried best to explain it but if ny doubt prevails pls tell me.
Regards
Gurminder
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Re: [Asterisk-Users] Call waiting setup/Confenencing problems in AAH
Hi, Sorry I can't help you in your questions but actually I have one. I m using TDM22B card. I am in india. I want to know are you able to get callerd ID? What cidsignalling you have set for in zaptel.conf.? On my system when a call comes it checks for caller ID and returns and error. #error ERROR[12656]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-24) # I am not able to understand what it is? Do tell me how yours is working. I am working in Indian Institute of science, Bangalore Gurminder On 8/30/05, Raj, Ashok [EMAIL PROTECTED] wrote: Hello I have couple issues with AAH. 1.5 1. Flash panel doesn't show proper status. Sometime accessing with IP seems to work and it shows current line status etc. Sometimes accessing with hostname of the asterisk server seems to show lines, but it doesn't show off hook etc when we pickup a extension and talk. In /var/www/html/panel/op_server.cfg I have tried setting manager_host to all possible values. 127.0.0.1 and its own ip address or its hostname. I have tried to reload with asterisk -rx reload, and also a system reboot, none help to get FOP working properly. 2. Call waiting. - Does the default configuration disable call waiting? I remembered with the same setup when I call myself with X-lite, I used to have an incoming call at line3. Now I get forwarded to the busy message. Any idea how I can get call-waiting to work? 3. Do we need special hardware to conference? I tried pulling an extension to an already in progress call, but it asks for a password. Don't know which one of the default passwords would work. Is there a default password we need to set? Cheers, ashok raj - Open Source Technology Center ___ --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] Caller ID
Hello, I am implementing asterisk in my office in India. I am facing a problem of caller ID. Any kind of help is appreciated. As a call comes to asterisk console it shows ERROR[27863]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-7) WARNING[27863]: chan_zap.c:5434 ss_thread: CallerID feed failed: Success Aug 18 11:33:45 WARNING[27863]: chan_zap.c:5476 ss_thread: CallerID returned with error on channel 'Zap/4-1' Thankyou Gurminder ___ 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: how to set the voice message as email attachment ?
I think you can add these lines in voicemail.conf
emailsubject=[VMBOX]:New message ${VM_DATE}
emailbody=Hello ${VM_NAME}:\n\tYou have a new voice
message.\n\tMessage Duration: ${VM_DUR} mins\n\tCaller ID:
${VM_CALLERID}\n\t( !)\n\t Date: ${VM_DATE}. \nThanks!\n--The Netlabs
SoftCall Service\n
filename='voicemail'
attach=yes
saycid=yes
sendvoicemail=yes
review=yes
operator=yes
delete=yes
Do tell me it works
Bye
Gurminder
On 8/11/05, larry lin [EMAIL PROTECTED] wrote:
Hi there,
I am using redhat 9.0 with asterisk 1.0.7.
I created an user and was be able to leave voice messages to that user
and retrieve the voice message. I looked the wiki and setup the voice
message as the email attachment. However, I have never received email with
the voice attachment. Here is the setting for voicemail.conf:
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
;format=wav49|gsm|wav
format=wav
; Who the e-mail notification should appear to come from
[EMAIL PROTECTED]
;[EMAIL PROTECTED]
; Should the email contain the voicemail as an attachment
attach=yes
;Turn on/off envelope playback before message playback. [ON by default]
envelope=yes
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M
'hours'
[default]
3114 = 3114,larry lin,[EMAIL PROTECTED],,attach=yes
Do I missing anything ? Thanks in advance.
Larry
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Re: [Asterisk-Users] info regarding hardware
Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit [EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance, Ankit P.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ 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] info regarding hardware
I m using it on POTS line and will start with ISDN soon :-). Cheers Gurminder On 8/9/05, Ankit [EMAIL PROTECTED] wrote: hi gurminder, are you using it on isdn line or pots line? On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote: Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit [EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance, Ankit P.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ 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 ___ 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 call from iax extn but cannot call it - unable to cteate channel iax
Hi Angus,
If you can send your general settings of iax.conf may be
I can work it out.
Regards
Gurminder
On 8/7/05, Angus Comber [EMAIL PROTECTED] wrote:
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid=some name 300
auth=md5
Then in my extensions.conf I have:
exten = 300,1,Dial(IAX/${EXTEN},20)
exten = 300,2,Hangup
I can dial from iaxComm (a soft IAX client) and that works fine. But when I
try to dial 300 get:
WARNING[22077]: channel.c1970 ast_request: No channel type registered for
'IAX'
NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of type
'IAX'
I have restarted Asterisk after config change.
What have I not done. I am just testing the iaxComm program.
Angus
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[Asterisk-Users] CVS not responding
Hi All, I am trying to connect to cvs.digium.com but connection gets timed out. Even pinging to cvs.digium.com is not working. I m using cvs login password - anoncvs Regards Gurminder ___ 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] what is the problem with gmail and the list.
On 8/2/05, John Novack [EMAIL PROTECTED] wrote: Rich Adamson wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. I suspect that any dump the list might or might not have taken isn't the complete story. m using gmail and suddenly got all the mails after 29th july... on 2nd august. --gurmi John Novack ___ 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
