Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p
The ast-UK-and-DTMF-pol-CID.diff patch will only work for the tdm400 FXO modules due to the fact that the x101p is not capable of detecting polarity reversal events. Dan On Fri, 2004-09-10 at 17:38, Robert Boardman wrote: should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009 (ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I backed up to cvs as of 31/08/04 and that worked fine. Zapata.conf: usecallerid=yes cidsignalling=v23 cidstart=polarity usecallerid=uk doesn't work, has this changed somewhere along the way, or is this something else? Caller ID detects fine, although I get this logged to asterisk console: Sep 6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I'll try and add this to the wiki when I get time Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 06 September 2004 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ 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 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] UK Disconnect supervision with TDM400P
On Wed, 2004-09-01 at 22:02, Edward Eastman wrote: Hi, thanks for the reply, only just got round to having a look at it again (annoying how real life gets in the way of the important stuff ;) I've had a go at ramping up the tx/rx gain but it doesn't seem to make any difference. FWIW it's the same with the module in normal fcc mode. Does anyone know if bt do normally provide disconnect supervision or whether it has to be done with e.g. busydetect (and can either be detected by the tdm400p in uk mode)? Thanks Ed Edward Eastman wrote: I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the remote party hangs up, I get approx 20secs of the bt line hungup tone before asterisk hangs up, which leads (if nothing else) to the well documented 20secs of beep on vm problem :) I have tried: busydetect=yes / busycount=7 / other busycounts / callprogess=yes but none of these make any difference. I have loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks signalling. Try increasing your RX gain in 1db steps, until it reliably hangs up. I had a box with X100Ps which busydetected perfectly with default gain settings. When they were replaced with TDM FXOs, busydetect stopped working and I needed 3db of RX gain added to get it working again. Regards, Richard Ed, When someone does hang up on you with your BT line, what do you hear? Here I get a click/pop following by a 4 second unobtainable tone followed by a click/pop... The clicks are BT's 'k-break's... It obviously doesn't seem to be what * expects... Investigating this is something I'm hoping to have a look at soon, but, if you have time beforehand BTW have you used the IRC channel? Dan (dant) ___ 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
; ; 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 On Wed, 2004-02-04 at 14:44, Deepakumar JV wrote: 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 ___ 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] IAX call problems
Hi Rattana, Do you have jitterbuffer enabled? Dan On Fri, 2004-01-30 at 13:40, Rattana BIV wrote: hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 1 [192.168.1.22:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 65795ms SCall: 21588 DCall: 6 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: PING Timestamp: 75906ms SCall: 22105 DCall: 5 [192.168.1.77:4569] Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 75906ms SCall: 5 DCall: 22105 [192.168.1.77:4569] Any suggestions ??? Thanks in advance Rattana PS: The softphone I use work with wiax.dll and is developpe by me =) ___ 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] Re: Cisco SIP license?
The £ came through here OK... --- These optional licenses (which can also be purchase separately, and are approx £10/$15) are to upgrade the number of users on the Cisco Call Manager Platform. --- Dan (in UK) On Sat, 2004-01-03 at 13:34, Adthrawn wrote: In case anybody is trying to work out the currency I used - it's actually British Pounds, but the £ sign isn't being handled by the mailing list. I've noticed that the mailing list is also having problems removing the HTML or Microsoft OLE email components, and is constantly filling the list with the background gunk. It seems to also have a problem with certain platform's line breaks. It's not affected my other emails apart from this last one though... Bizzare... Ad. On 3 Jan 2004, at 5:22 am, [EMAIL PROTECTED] wrote: These optional licenses (which can also be purchase separately, and are=20= approx =A310/$15) are to upgrade the number of users on the Cisco Call=20= Manager Platform. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Avaya IP phones
On Thu, 2003-12-04 at 23:02, Ed Rubright wrote: The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether its a SIP phone or not. Thought maybe someone on this list would know. It's not SIP, currently, well over a year ago Avaya demonstrated SIP functionality in both the phones and in their Multivantage PBX software. This has not been released, apparently due to lack of business demand... I've tried making one work as is with asterisk and a number of other h323 products, however, I've not yet had any success... Avaya seem to have really filled these with lots of proprietary hacks... Question: Would I be able to register my Asterisk server or an individual SIP phone (Cisco 7960 or Polycom IP600) with the Avaya server these 46xx IP telephones use? I don't know what model of the Avaya server the company has purchased, so I have limited info here. You would not be able to get a SIP phone talking to the Avaya PBX for the reasons mentioned above... You could possibly get an h323 trunk between the * and the Avaya PBX, but, I have not tried this yet so this too may not work... Also, something worth mentioning, the number of IP trunks you can have is limited by the number the Avaya PBX is licenced to use... If your company is not currently using trunks, there may not be any trunks available to use... Not a lot of good news there, but I hope it's helpful to you... Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
