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Does anyone have any opinions on the music on hold quality over G729? The stock files seem to sound terrible over it, this is enhanced further by calls coming from the PSTN via a Zaptel gateway. I am only using the stock wav files and have not attempted to use much else so far. I've ruled out timing issues on the system generating the MOH itself (ztdummy on the PBX itself, our Zaptel gateway

Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. -------------- next part -------------- An HTML attachment was

Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23

Question was off topic for the thread, but I'm feeling helpful today. More of a 1234... make install modprobe usb-uhci modprobe zaptel modprobe ztdummy -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brad Sumrall Sent: Monday, March 19, 2007 13:17 To: 'Asterisk Users Mailing List - Non-Commercial

November? It's DD/MM/YYYY in his case, not MM/DD/YYYY. Either way, even two days is more than enough for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of john beaman Sent: Tuesday, April 03, 2007 12:43 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: asterisk-users Digest, Vol 33,

Look at app_valetparking here: http://www.voip-info.org/wiki/index.php?page=Asterisk+addons ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Kiely Sent: Monday, July 16, 2007 16:47 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MultiParking

Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extension at remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the users.

I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want to write a whole new parser to do it (especially not one that understands templates and

...s of beer (David Cook) 2. DUNDi, So Easy A Caveman Could Do It! (JR Richardson) 3. Polycom behind NAT won't register to * server behind ALG (Matthew Warren) 4. Re: Polycom behind NAT won't register to * server behind ALG (Alex Balashov) 5. Re: Polycom and NAT (Darryl Dunkin) 6. Re: Polycom behind NAT won't register to * server behind ALG (Henry L.Coleman) 7. Re: Polycom behind NAT won't register to * serverbehind ALG (Marty Mastera) 8. rfc3680, reginfo+xml (Olivier) 9. How to re-read values from database in Trixbox (Edgar Guadamuz) 10...

From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x10000 error Any Ideas? 1005195711|so |4|00|---------- Initial log entry ---------- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry

Hi, We have a PRI connection & when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats

...ep 2001 12:21:22 -0700 (PDT) Received: by www.postofis.com (Postfix, from userid 0) id C41EC18C3B3; Wed, 5 Sep 2001 14:23:56 -0500 (CDT) Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 X-Mailer: MIME::Lite 2.117 (F2.6; B2.11; Q2.03) From: Dunkin Donuts & Coca Cola <dunkindonuts_cocacola@postofis.com> To: samba@samba.org Subject: TANKER COLA alana, yanýnda BEDAVA DONUT Message-Id: <20010905192356.C41EC18C3B3@www.postofis.com> X-Converted-To-Plain-Text: from multipart/mixed by demime 0.98b X-Converted-To-Plain-Text: Alternati...

You have it right, for 1.2, use 'tos=', for 1.4 use 'tos_sip/tos_audio/tos_video'. ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Al lists Sent: Monday, July 23, 2007 10:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fwd: Asterisk and COS

In your per-phone configuration: <phone1> <reg> ... <divert divert.fwd.1.enabled = "0" divert.fwd.2.enabled = "0" divert.fwd.3.enabled = "0" divert.fwd.4.enabled = "0" divert.fwd.5.enabled = "0" divert.fwd.6.enabled = "0" /> This removes the soft-key and disallows the