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May be I have fiddled too much with my sipura settings but I can't get it to give the stutter tone when there is a new voice mail waiting on the asterisk box. I can either get a stutter tone all the time or not at all. Anyone got this working. Thanks Chris

Looks like half a patch has been applied to app_dial in cvs head could someone with commit rights fix it. Thanks Chris

Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. Chris

Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris

...68.123.20 for seqno 102 (Critical Response) Any ideas I am sure it was working ok with cvs head a month ago. Chris ---- sip debug ---- Retransmitting #7 (no NAT) to 192.168.123.20:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.123.20:5060;branch=z9hG4bK49428a54 From: "Chris Stenton" <sip:201@pbx.gnome.co.uk>;tag=000b46a0866100290e5a9947-05a960d9 To: <sip:607@pbx.gnome.co.uk;user=phone>;tag=as4d62d2f2 Call-ID: 000b46a0-8661000a-4405e325-7e25031f@192.168.123.20 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE...

Anyone written an extension that will take a 12 digit number, convert it to an IP address so that you can make a sip call to it. Chris

I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @<proxy-addr>... So the callerID on the phone looks like: 2145551212@10.10.10.10 which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? 7.5

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing What have you set the PSTN Dialing Delay: on the PSTN Line tab (logged in as admin advanced) ? M...

I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Does anyone have experience setting up * to accept anonymous sip UAs and the dumping the call into IVR? I'm thinking this would be a good way to have customers call us without creating an extension. So for my tests have been focused on providing internal functionality. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ews CURRENTLY: Asterisk and libpri in Asterisk-current (CVS head) build and run on FreeBSD 5.2.1. The last major change (support for multiple CPUs) has been incorporated into Asterisk-current. Asterisk should now be thread safe on FreeBSD and testing on dual CPUs has just begun, thanks to Chris Stenton [Thanks Chris!] Zapata and zaptel in Asterisk-current have not yet been ported. The zaptel driver has been enhanced significantly since the 0.9.0 version present in FreeBSD's ports. So, the Asterisk application will not build or run with FreeBSD's zaptel 0.9.0 as it stands. CHANGES THIS...

There was some discussion on this list recently about the voiptalk silver service. I've just had an e-mail from them saying that the price has been reduced to 2.99 per month. However, they still only provide an 0870 number whereas pipecall provide a local call rate 0845 number in the fee. Chris

Ok I'm no cvs expert is there a cvs command to get a date sequential cvs log archive for cvs-head or a URL for it. With so many daily changes its hard to keep track of what the changes are. Thanks Chris

I can't get my 7960 to upgrade to 7.2 or infact downgrade to 6.3 either. when I change image_version to the 7.2 string in sipdefault.cnf it reboots but does not look for any image files it just looks for the same files as it normally does. Any ideas? it seems as though I have locked the firmare. Chris

I've just upgrade to the latest head (20843) and I get the following error .Apr 17 08:41:07 WARNING[8527]: loader.c:726 __load_resource: new style cdr_pgsql.so (0x0) loaded RTLD_LOCAL Apr 17 08:41:07 WARNING[8527]: loader.c:744 __load_resource: Key routine returned NULL in module /usr/lib/asterisk/modules/cdr_pgsql.so Apr 17 08:41:07 WARNING[8527]: loader.c:753 __load_resource: 5 errors

anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT) thanx very much __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

...e) > 7. Re: permission problem (Wolfgang S. Rupprecht) > 8. Re: New Asterisk bounty: SIP simultaneous (Sunrise Ltd) > 9. Re: Gogoif with variables acting funny? (Shaun Dawson) > 10. Re: PRI numbering plan (Martin List-Petersen) > 11. Re: X101P FXO with RED alarm (Chris Stenton) > > --__--__-- > > Message: 1 > From: "Steve Totaro" <asterisk@totarotechnologies.com> > To: <asterisk-users@lists.digium.com> > Date: Sat, 12 Jun 2004 11:40:07 -0400 > Subject: [Asterisk-Users] Changed IP and subnet now no SIP Register 403 > Rep...

The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension number&password could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there