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Has anyone else had problems with users being able to press key tones during a voice prompt? I have a few users complaining that some systems will not recognize key presses during them. using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode. Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

...missed ? Any idea to help me ? What can I describe or search more for a best analyze ? Many thanks in advance, guys ! Best Regards, Francois BERGERET, France. ------------------------------ Message: 22 Date: Tue, 27 Sep 2005 09:25:20 +0200 From: "Andreas Sikkema" <andreas.sikkema@bbned.nl> Subject: RE: [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <5B97FFB7F28E3E4294B049C968CBF47B5C99ED@SERVER-EX02A.INTERN.BBNED.NL> Content-T...

...l rates but I fully expect wholesale > rates based on call prefix will be available for carriers / ITSP In some countries there's a company (companies?) providing access to a database which telcos can use to find the rates on this kind of numbers. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

Hi, Has anyone written a SPEC file for zaptel, with kernel 2.6 and udev support? I can find some spec files here and there, but from what I can see they're all kernel 2.4 / non udev... -- Andreas Sikkema bbned NV Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

...is not recorded (0 byte file is created) and it gives the > following errors - > > "unable to convert from g729 to slin" You can force Record to record to G.729, but I'm not sure the voicemail application has the same possibility. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

I have set up extension.conf and sip.con with default parameter of UNIVOICE server, but Asterisk show this message when I call a number: Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899 create_addr: No such host: univoice,Ttr Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is

...is configured to use rfc2833 but is also a little to lote with the filtering out of the DTMF. So sometimes it's not Asterisks fault at all ;-) And then there's some IVR's that don't notice it at all, while others are totally unusable. -- Andreas Sikkema BBned NV Software Engineer Planeetbaan 4 +31 (0)23 7074342 2132 HZ Hoofddorp

Hello all, I want to sniff all these info to test a sip ip phone talking to a asterisk server. I have used tcpdump, but It just shows the UDP, length: 602 Anyway to see the sip uri. Host info? Regards, Dinesh. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them

Can anyone comment or share experences with using Dell 2850's with Asterisk. Proposed config is 2850, 2 x 3.6g procs, 2 g's of ram, 4 x 36g 15k rpm drives raid 10, Digium TE411P ( the echo cancelling cards ). Expected load is 1 or 2 pri's (most likely 1 ) 100 Polycom phones on the local network, 15 phone on a remote T1. 6 phone remote via the internet using IAX, Voicemail for

I am trying to enable dial-by-email by using LDAPget to query an Active Directory server. I've got it retrieving the phone number fine. Unforunately, the numbers stored in active directory are either in the format: (xxx) xxx-xxxx or xxx-xxx-xxxx. Is there any way to parse characters out of the dialed phone number so that I only end up with digits (remove spaces, parenthesis and dashes)?

Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Thanks, Doug. -------------- next part -------------- An HTML

Hello all, I really appreciate the replies I've gotten about this so far (especially the support for wanting to run it on Solaris!). The core issue seems to have been missed, though -- is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? This probably comes down to a few issues: - Is ztdummy (a component of Zaptel) *really*

I'm publishing tftp through my firewall to support external Cisco 7960 sip phones. I know that the primary port is 69 for tftp. However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range) In an effort to limit the secondary ports that are opened, some Windows based tftp server such as the winagents product allows you to limit the range of secondary ports that are

Hi, Occassionally Asterisk will go down and I have to restart it.. not often.. but sometimes. When it does the manager interface stops working, as does the CLI. My thoughts was to poll the manager interface once every 5 minutes for a value. If I don't get the value back then alert me that the server is possibly down. Does anyone know what a good value to poll for might be? I was thinking

...255.255.255.0 U 0 0 0 eth1 192.168.2.0 * 255.255.255.0 U 0 0 0 eth1 82.161.58.0 * 255.255.254.0 U 0 0 0 eth0 192.168.0.0 * 255.255.0.0 U 0 0 0 fourdigits default bbned-10k-07.ro 0.0.0.0 UG 0 0 0 eth0 Host "50K" tinc-up: echo "1" > /proc/sys/net/ipv4/ip_forward ifconfig $INTERFACE 192.168.1.1 netmask 255.255.0.0 Host "Oeoe" tinc-up : ifconfig $INTERFACE 192.168.2.1 netmask 255.255.0.0 ================ --...

...phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not playing the recorded sound from your microphone back to your loudspeakers. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models?