[Asterisk-Users] prepaid app
Hello Has anyone played with the prepaid app available at http://www.voip-info.org/tiki-index.php?page=Asterisk+callingcard? I have setup the postgresql database with required tables and records. I am facing this problem when i enter the card details WARNING: Error occurred while executing PL/pgSQL function asterisk_authenticateWARNING: line 10 at select into variables Any idea? Regards Deepak
[Asterisk-Users] oh323 codec negotiation
Hello I had this codec negotiation with oh323 call. i used G723 codec and the provider had G729 as first priority. In this situation what ever number i dial i used get "No one there to answer the call". As soon as i changed my codec to G729 the call went through but had other problems, which i got away by dowloading the latest code for oh323. Has anyone seen this problem? or it is normal? Regards Deepak
Fw: [Asterisk-Users] voicemail extension - hangup
Hello friends Is there any way to run a program or dial command after the user hangsup from voicemail or voicemailmain ? the control does'nt come back to the next priority instead the call ends within voicemail/voicemailmain. Any ideas? Regards Deepak - Original Message - From: Deepakumar JV To: [EMAIL PROTECTED] Sent: Monday, February 16, 2004 01:22 PM Subject: [Asterisk-Users] voicemail extension - hangup Hello, I have configured 8500 to access voicemailmain. With whatpriority does the control exit when the user hangsup the phone without pressing #. I want to execute an app when the control exits from voicemailmain. Any inputs? Regards Deepak
[Asterisk-Users] voicemail extension - hangup
Hello, I have configured 8500 to access voicemailmain. With whatpriority does the control exit when the user hangsup the phone without pressing #. I want to execute an app when the control exits from voicemailmain. Any inputs? Regards Deepak
Re: [Asterisk-Users] Digium connectivity issue?
I am having same problem and i was never successful in connecting to digium.com or asterisk.org or asteriskpbx.org for last three days. Deepak - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Friday, February 13, 2004 01:54 PM Subject: [Asterisk-Users] Digium connectivity issue? Are others seeing hugh delays and/or lack of connectivity to Digium? Rich ___ 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 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] DTMF over SIP to a Cisco gateway
Hi, I am trying to use the voicemail feature of * for a Cisco call manager express setup with 10 7960 phones. * is unable to recognise the DTMF when mailbox is accessed by voicemail. Here are my configs in cisco dial-peer voice 8500 voipdestination-pattern 8500session protocol sipv2session target sip-serverdtmf-relay rtp-ntecodec g711alawno vad in extension.conf exten = 8500,1,VoicemailMainexten = 8500,2,Hangup What am I missing here? Regards Deepak
[Asterisk-Users] incoming DTMF on a SIP call
Hi, How do i set the DTMF mode for incoming SIP call per context ? Or is there a global config that i can set for all context? I am haing trouble getting the DTMF tones from a cisco router with rfc2833 mode. when i make a call from 7960G via a 3640 (cisco call manager express) to asterisk to check voicemail, i am unable to key in any number because * does not understand teh DTMF. I am not sure whether it is a * configuration or cisco configuration. Any ideas? Regards Deepak
[Asterisk-Users] H323 calls via provider
Hello I am trying to use a VOIP provider (PC to PSTN). Is it possible to use asterisk as a client and make calls via a H323 provider? Can anyone guide me how the oh323.conf should be and extension.conf should be. I have a IP, userid and password given by them. I am using www.mywebcalls.com. Has anyone tried using * like this? Regards Deepak
Re: [Asterisk-Users] talking clock
How about a followup post showing exactly what your extensions.conf
entries look like, and what you had to go to get it twekaed to your
satisfaction?
Here is the working extension.conf i came up with
[time]
exten = 5559,1,Answer()
exten = 5559,2,Playback(time)
exten = 5559,3,SayUnixTime(||IM)
exten = 5559,4,SetVar(TIME1=${DATETIME})
exten = 5559,5,SubString,TIME2=${TIME1}|-2|2
exten = 5559,6,Playback(beep)
exten = 5559,7,SayNumber(${TIME2})
exten = 5559,8,Playback(second)
exten = 5559,9,Wait(1)
exten = 5559,10,Goto(time,5559,2)
but then i got to know about the S option in SayUnixTime() from Dan. THANKS
DAN.
exten = 5558,1,SayUnixTime(|GB|IM 'beep' S 'second')
exten = 5558,2,Goto(time,5558,1)
Thanks to everyone for helping me.
Now i have small problem which i am trying to fix with my less programming
knowledge. I get to hear the time in odd intervals, like 11:30:06 then
11:30:11 then 11:30:15 then 11:30:19 then 11:30:19 so the interval varies 4
and 5 seconds alternatively.
I wanted this clock to tell the time every 10 seconds and it should be the
actual system time.
ie., at 11:30:20 it should execute 5558,1 and at 11:30:30 it should execute
5558,1 that way i can hear the time every 10 seconds.
Regards
Deepak
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[Asterisk-Users] talking clock
Hello I am looking for a AGI application that can say the current time with seconds, but i don't need the day/year. Has anyone got this already? Thanks in advance Deepak
Re: [Asterisk-Users] talking clock
Thanks for your reply Brian. I am able to get only the hour and minute but not the seconds. I need seconds also, any suggestions? Regards Deepak - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 02:23 PM Subject: Re: [Asterisk-Users] talking clock SayUnixTime will do that just give it the format you want. SayUnixTime([unixtime][|[timezone][|format]]) unixtime: time, in seconds since Jan 1, 1970. May be negative. defaults to now. timezone: timezone, see /usr/share/zoneinfo for a list. defaults to machine default. format: a format the time is to be said in. See voicemail.conf. defaults to ABdY 'digits/at' IMp Returns 0 or -1 on hangup. bkw On Wed, 4 Feb 2004, Deepakumar JV wrote: Hello I am looking for a AGI application that can say the current time with seconds, but i don't need the day/year. Has anyone got this already? Thanks in advance Deepak ___ 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 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] talking clock
Thanks to everyone. I got the talking clock working the way i wanted. thanks again Deepak - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 03:20 AM Subject: Re: [Asterisk-Users] talking clock At 11:50 PM + 2/4/04, Dan Tucny wrote: ; ; Talking clock (123) ; exten = 123,1,SayUnixTime(|GB|HM 'vm-and' S 'digits/seconds') exten = 123,2,Wait(1) exten = 123,3,Goto(1) the seconds sound can be picked up from John Todd's site, http://www.loligo.com/asterisk/ Dan [snip] The file seconds.gsm is also in asterisk-sounds, which along with many other interesting and amusing clips can be pulled from the CVS server just like asterisk, zaptel, etc. JT ___ 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 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] sipphone dialing out problem
Hello
when i dial a toll free no using sipphone
i get this error message. How do i solve this?
Any help will be appreciated.
console message:
Starting simple switch on
'Zap/2-1' -- Executing SetCallerID("Zap/2-1",
"17473863282") in new stack -- Executing
SetCIDName("Zap/2-1", "Deepak JV") in new stack --
Executing Dial("Zap/2-1", "SIP/[EMAIL PROTECTED]")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/proxy01.sipphone.com-8efc is making progress passing it to
Zap/2-1 -- SIP/proxy01.sipphone.com-8efc answered
Zap/2-1Feb 3 22:15:57 NOTICE[1218901440]: channel.c:1481
ast_set_read_format: Unable to find a path from G729A to ULAWFeb 3
22:15:57 NOTICE[1218901440]: channel.c:1451 ast_set_write_format: Unable to find
a path from ULAW to G729AFeb 3 22:15:57 WARNING[1218901440]:
chan_zap.c:3728 zt_write: Cannot handle frames in 256 format == Spawn
extension (internal, 18006526672, 3) exited non-zero on
'Zap/2-1' -- Hungup 'Zap/2-1'
Regards
Deepak
[Asterisk-Users] echo cancellation disabled
Hello I get these entries in my event log Jan 31 19:21:08 gateway kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Do I have to change anything for enable echo cancellation? Regards Deepak
[Asterisk-Users] billing software
Hello Is anyone using a commercial billing software with * which product is that? i am looking for using with pre-paid as well as post paid. Also where can i find info about voip regulation/licenses to become a provider??? Thanks Deepak
[Asterisk-Users] specific to X100P with UK telephone lines
Hello all,
I got this wierd problem with X100P. When
i try to dial any no over the PSTN line, i get only the dial tone.
extensions.conf
[landline]
exten =
_9.,1,Dial(Zap/1/${EXTEN:1})
indications.conf
[general]country=uk
zapata.conf
[channels]usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallprogress=nocallwaitingcallerid=yesthreewaycalling=notransfer=nocancallforward=yescallreturn=yesmailbox=02082860552echocancel=yesechocancelwhenbridged=yesechotraining=yes
txgain=0group=1callgroup=1pickupgroup=1;callerid=asreceivedamaflags=documentation
signalling=fxs_kscontext=incomingimmediate=yeschannel =
1
callerid=02082860552echocancel=yessignalling=fxo_kscontext=internalimmediate=nochannel
= 2
Is there any specific settings that i need
to do to use X100P card with UK telephone lines? Telewest is my service
provider.
Is anyone using X100P in UK with telewest
without any problem? could you share your settings or give me some
direction?
I approached digium on this and got a RMA
X100P card also, still the same problem. Tried in a different system also , same
problem. Wondering what would be the cause??
Regards
Deepak
[Asterisk-Users] festival patch missing in latest CVS or stable build
Hello I downloaded the stable build of * and was not able to find the festival patch in that build. Also i tried from CVS and the same. Can anyone tell me where i can find the festival patch? Or the patch is no more requried. i can just compile festival and it will work?? Thanks in advance Regards Deepak
[Asterisk-Users] IAX2 / SIP testing
Hello I am interested in testing the voice quality with another * setup in India. I am planning to setup a * server at India and would like to know whether the audio quality would be good enough to make freequent calls. anyone willing to help please let me know and i can test in your convenient time. Thanks in advance Regards Deepak
[Asterisk-Users] IAX2 / SIP testing
Hello I am interested in testing the voice quality with another * setup in India. I am planning to setup a * server at India and would like to know whether the audio quality would be good enough to make freequent calls. anyone willing to help please let me know and i can test in your convenient time. Thanks in advance Regards Deepak
[Asterisk-Users] festival patch missing in latest CVS or stable build
Hello I downloaded the stable build of * and was not able to find the festival patch in that build. Also i tried from CVS and the same. Can anyone tell me where i can find the festival patch? Or the patch is no more requried. i can just compile festival and it will work?? Thanks in advance Regards Deepak
[Asterisk-Users] SIP error
Hello When ever i make calls via a SIP provider I keep getting this error message Jan 29 02:09:20 NOTICE[1228887360]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible any idea what is it? Regards Deepak
[Asterisk-Users] PSTN gateway
Hello Has anyone come across a small residential PSTN gateway? Its not worth running a * just as a PSTN gateway as it requries a seperate system / power / etc... I am looking for a device that could connect to * and a pstn line so that i could register that device to * and make pstn calls via that device. Regards Deepak
Re: [Asterisk-Users] PSTN gateway
Sorry for confusing.. let me explain ideally i want to have two * running, one at my place and the other at a remote location. Now the problem in running * at a remote location is the effort / cost involved in setting up / maintaining the * box. Hence i was looking for a device that could register with * (as a client so that i could dial a number and reach it as a normal extension) and also have a PSTN connectivity at the remote location. The reason i need PSTN connectivity at remote location is to make outbound calls from * via the device so called PSTN gateway. If i am still not clear, then in simple terms, i am looking for a hardware device with one FXO port and SIP support. Any help or suggestion please Thanks in advance Deepak - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 01:58 PM Subject: Re: [Asterisk-Users] PSTN gateway - Original Message - From: Deepakumar JV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 8:37 AM Subject: [Asterisk-Users] PSTN gateway Hello Has anyone come across a small residential PSTN gateway? Its not worth running a * just as a PSTN gateway as it requries a seperate system / power / etc... I am looking for a device that could connect to * and a pstn line so that i could register that device to * and make pstn calls via that device. I'm confused. Do you want to get rid of *, or not? It sounds like you're just looking for an IP phone to pstn gateway service. See: vonage, voicepulse, etc... - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ 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 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] OH323 config file format
Hi Tom Thanks for your reply. SIP: I am using for my regular international calls, hence i have not tested with multiple calls. i have never experienced any crash. its quite stable and gives good performance. my system config is pentium III 300Mhz with 64MB. OH323: I am trying to register with www.mywebcalls.com. My basic understanding of H323 is, since * can act as a client (H323) as well as a gateway (H323) I should be able to use any of the H323 providers (even if they provide they own H323 based clients or web based H323 clients). So this provider has given me a username and password, that's all.. nothing else. Since they provide a web based client, they don't support any other configuration i am unable to get more details from them. But by using their web based client I have found out the IP of (i am not sure whether its a GW or GK). I am not sure where to enter the username and password that they have given. Thanks again Deepak - Original Message - From: T. Chan To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 10:00 PM Subject: RE: [Asterisk-Users] OH323 config file format Hi, Deepak, how are you? I don't quite understand what you meant by username and password sending calls to a H323 service provider, do you mean you have to register onto their gatekeeper? Or otherwise, you should not need username and password. Meantime, I am trying to setup up SIP calling to a service provider, can you let me know what is the maximum number of calls you have experienced with sending SIP calls to your service provider? Have you experienced any crash? What is the configuration of your computer? Thanks Tom -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Deepakumar JVSent: Wednesday, January 21, 2004 12:38 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] OH323 config file format Hello I am trying to configure my extensions.conf and oh323.conf to termination calls to a H323 service provider. Can anyone send me a sample config files? or tell me where to put the username and password which my service provider has given? also how to put the Dial command in extensions.conf Asterisk rocks. I have a SIP provider configured for all my international calls and it works absolutely fine. Its cool. Thanks in advance Regards Deepak
[Asterisk-Users] OH323 config file format
Hello I am trying to configure my extensions.conf and oh323.conf to termination calls to a H323 service provider. Can anyone send me a sample config files? or tell me where to put the username and password which my service provider has given? also how to put the Dial command in extensions.conf Asterisk rocks. I have a SIP provider configured for all my international calls and it works absolutely fine. Its cool. Thanks in advance Regards Deepak
[Asterisk-Users] fwd problem with *
Hello I am trying to register for fwd from * but having problem and unable to solve it. I keep getting this message *CLI NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as62a7f29b'NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as62a7f29b'NOTICE[1125329600]: File chan_sip.c, Line 2991 (sip_reg_timeout): Registration for '[EMAIL PROTECTED]' timed out, trying again my sip.conf [general]port=5060disallow=allallow=gsmallow=ulawallow=alawcontext=4919maxexpirey=180defaultexpirey=160tos=reliabilityregister=89699:[EMAIL PROTECTED]/4918 [fwd.pulver.com]type=peersecret=aimccieusername=89699host=fwd.pulver.comauth=plaintextfromuser=89699 my extension.conf [general]static=yeswriteprotect=no [globals]CONSOLE=Console/dsp [default] include 4918 [4918]exten = 4918,1,Dial(SIP/4918,15,t) ; see "show application dial" for options and formatsexten = 4918,2,Voicemail2(u4918) ; go to Voicemail2 if phone is "U"nansweredexten = 4918,102,Voicemail2(b4918) ; go to Voicemail2 if phone is "B"usyexten = 4918,103,Hangup ; and then hangup Thanks Deepak
[Asterisk-Users] asterisk with a third party gateway
Hello Can asterisk be configured as a PBX with a third party gateway (cisco router 3640 running Cisco call manager express). The cisco gateway will only interface the PSTN and asterisk, so the cisco routerwill handle incoming and outgoing calls. I would like to do this as we have the hardware like VIC ISDN cards with us. The main reason for going with asterisk is to hook up with a VOIP service provider. Any help in this regard will be greatly appreciated Thanks Deepak
[Asterisk-Users] voicemail
Hello Has anyone implemented a voicemail system for an existing cisco call manager express VOIP setup. I am looking for a open source based software which can integrate with my existing Cisco call manager express VOIP (based on 3640 router) with individual users having their own voicemail box. Thanks, Deepak
