RE: [Asterisk-Users] Billing

Maxim, based on the info in the URL below, you claim to say that completely asterisk based solution for calling card application may not scale. You suggest that the alternative is to use gnugk just to use its AAA, or Radius. In my opinion and experience, I would say by introducing Gnugk and OH323,

[Asterisk-Users] SIP to H.323 no audio

Hi, I am trying to make a call from SIP to H.323 using chan_h323. Asterisk CVS-HEAD-03/10/05-10:08:22. As given in chan_h323 readme I compiled pwlib and open h323 versions 1.8.1 and 1.15.1. Call seems to be get connected but no audio path. I see following; -- AGI Script Executing Application:

[Asterisk-Users] broadvoice patch and 16 second re-registers

Hi, With the Patch, now I see following log notices every 13-14 seconds on my console for each SIP provider. Nov 10 22:52:06 NOTICE[1089948224]: chan_sip.c:4023 sip_reregister:-- Re-registration for [EMAIL PROTECTED] Nov 10 22:52:06 NOTICE[1089948224]: chan_sip.c:6795 handle_response: Outbou

RE: [Asterisk-Users] iconnect incoming problems

great advice. Cheers Sathya > -Original Message- > From: Steve Rubin [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 10, 2004 6:55 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] iconnect incoming problems > >

RE: [Asterisk-Users] Broadvoice asterisk patch

Steve, My asterisk server is on public internet, since applying this patch I see my asterisk is sending re registration requests every 14 seconds. Nov 10 18:14:30 NOTICE[1089948224]: chan_sip.c:4023 sip_reregister:-- Re-registration for [EMAIL PROTECTED] Nov 10 18:14:30 NOTICE[1089948224]: c

[Asterisk-Users] iconnect incoming problems

Hi,   I cannot receive any calls via icoonect. I can make outgoing calls, and also I can see sipauth.deltathree.com registering me correctly (I am on public internet). When I try calling-in I wouldn't even get an invite my way. I then hookup a grandstream ata and without a problem it was abl

[Asterisk-Users] timing and dropped calls

Hi,   I have a * server which does only SIP to H323 completely in IP domain, there is no digium h/w in it. In your experience, for this type of application, is it required to have a timing source to prevent the calls being dropped.   Cheers   SW   _

RE: [Asterisk-Users] ATCC - Astcc-Admin.cgi File

Would somebody like to send me a few gsm files? I need the following: "please enter your pin number followed by the pound sign" "That pin number is not valid" what I do is use the free text to speech service at http://www.rhetorical.com/cgi-bin/demo.cgi. use the option download wav file, then you

RE: [Asterisk-Users] iconnect inbound - so do we know how to fix it

inbound - so do we know how to > fix it > > > May be you can find the solution in my post: > > http://lists.digium.com/pipermail/asterisk-users/2004-August/058014.html > > Raj > > --- Vladyslav <[EMAIL PROTECTED]> wrote: > > > Try to comment out in your

[Asterisk-Users] iconnect inbound - so do we know how to fix it

Just wondering whether we have a resolution to iconnect incoming problem,  which started few days ago.   Cheers SW

[Asterisk-Users] astcc help

Hi,   Really appreciate if someone who got astcc working lists the steps to make it work. I've got it installed and using the gui could get the database created. Would like to know how those two .conf files be populated and some pointers to the important fields in the database.   Thanks   Sa

[Asterisk-Users] new bounty for modifying calling card application to mysql

Hi,   I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL   Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ?   SW

[Asterisk-Users] how to integrate an addon module to asterisk

Hi friends,   Do we have some instructions to build an addon module to asterisk ?  Lets say I write my own addon module, addon.c, how do I go about linking this with asterisk. Any pointers ??    SW

[Asterisk-Users] codec pass-through feature

Hi Gurus, I we seen references to 'codec pass through feature' in the mailing list. SIP to SIP and SIP to chan_h323 as well. Could someone help me to understand this feature, or point me to some examples etc. Appreciate any pointers here. Thanks a bunch Sathya

[Asterisk-Users] Getting in to h323

Greetings, I am progressing well with this great product, the *. SIP to SIP calling, Vocal to *, Voicemail all in the past. Did Iconnect, FWD etc. Also, purchased couple of FXO cards and did zaptel as well. It's time to get to h323 now. Read the mailing list for H323 and OH323 etc. need some help

RE: Re: [Asterisk-Users] g723 to g723 SIP call - warning message

eems like * is trying to translate the codec. I have set G729 for both contexts. I thought if there is no codec translation, asterisk can handle pass through. Cheers Sathya > -Original Message- > From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 19

RE: Re: [Asterisk-Users] g723 to g723 SIP call - warning message

I am sorry I mean dtmfmode=info > -Original Message- > From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED] > Sent: Wednesday, November 19, 2003 10:34 AM > To: Eric Wieling; [EMAIL PROTECTED] Digium. Com > Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning mes

Re: [Asterisk-Users] g723 to g723 SIP call - warning message

Hi, Thanks Jeramy and Eric. Sorry for my ignorance. I still did not get the point. Do you mean that I have to set each of my context in sip.conf with dtmfmode=inband ? I have the GS phone set as DTMF mode = Via SIP Info. Would that need to be change to something else ? (Send DTMF:in-audio

[Asterisk-Users] g723 to g723 SIP call - warning message

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [EMAIL PROTECTED] sath]# cat

[Asterisk-Users] mysql addon

Hello, I am trying to install the cdr-mysql. Information given in the following kink is what I am trying to follow; http://www.voip-info.org/wiki-Asterisk+cdr+mysql I cant figure out where to install the asterisk-addons. Is it in /usr/src or /usr/src/asterisk ? Once I create the cdr-mysql.conf.

[Asterisk-Users] sending MWI to a none local client

Hi, I am using * to function as the voice mail system for Vocal. Since I do not have a context in sip.conf file for each vocal client, I can't set the mailbox= in sip.conf. How do I get the MWI to a Vocal client ? Cheers Sathya ___ Asterisk-Users

[Asterisk-Users] X100P - module does not gat loaded

Hi I installed a x100P card today. Once it is configured * no longer starting. It gives me the following error. == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074432736]: File chan_zap.c, Line 629 (zt_open): Unable to specify chan nel 1: No such device or address ERROR[1074432736]: Fil

[Asterisk-Users] call processing after a PIN

Hi folks, I would like some pointers to do a routine like this; 1 Call received at Asterisk 2 Asterisk answers with a voice prompt "enter a PIN number" 3 PIN received over DTMF 4 PIN is cross referenced in a database (SQL or FLAT file) 5 If a match is found, Asterisk prompt for "enter phone numbe

[Asterisk-Users] Read error on sound device

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Not

[Asterisk-Users] voicemail broken voice

> Hi, > > I am trying to set-up voice mail. > > When I dial the voicemail extension, voice prompt asking for > password is braking or intermittent. > > I see the error "File sched.c, Line 209 (sched_settime): Request > to schedule in the past?!?!. > > I am using a grandstream phone. > > BTW