search for: deillon

...Budgetone SIP phones. When I dial 555 (VoicemailMain), I hear "You have 5 new messages, 1- Read your messages, 2- , etc ... ) But when I dial 1 or 2 or everything else, nothing happen. Are they some lines wich do that asterisk listen my phones ? Thanks for your help, have a nice day Thomas DEILLON

Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have

Hi, I have an Alteon in test (a sip/rtp load balancer). This Alteon sends to the asterisk box a "SIP OPTIONS" to know if asterisk is alive. However, asterisk sends me a 404 message and not a response like, for example, a Thomson (200 + SDP) I wrote a very little script (you can find it at the end of the email) to send an Options message to asterisk/phones to try. It works

The file are there: http://thdei.info/results.zip and http://thdei.info/mos_6_MOS-USA_Test-114_20060605-042551cut-PESQ.png because, last time I put them in attachment and the mail was waiting for approvement and I never see it anmore . ________________________________ From: Deillon Thomas-WTD008 Sent: 05 June 2006 14:32 To: 'asterisk-users@lists.digium.com' Subject: Quality of Asterisk Hi, I have a problem with the quality test. So if you have a idea for me .... We test here, Motorola phone with Asterisk. Asterisk play sample to the mobile phone which record th...

...adeera. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/62 a82c69/attachment-0001.htm ------------------------------ Message: 17 Date: Wed, 7 Mar 2007 11:17:07 +0100 From: "Thomas Deillon" < Thomas.Deillon@smart-telecom.ch> Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com > Message-ID: < 86918CDC1242004D8B0563A43D1E2F0C027E2D09@e...

Hello, I make a sawdwish course in network and software engeneering at CPE lyon and in my company I'm working on Asterisk from 1 year. So, I'm looking for a internship (3 month) in a english country on a Asterisk project. Thanks, Thomas DEILLON

Hi all, I read somewhere that asterisk v 1.4 can make Stun and SNMP. I tried to find more information on these features but I didn't find any clues. Someone find a way to use it? Thanks, Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070122/f28cdeac/attachment.htm

Hi all, We are an ISP in Switzerland and we propose VoIP with Asterisk. Everything works perfectly for all clients but one. In a conversation, they have no sound during 2 to 8 seconds using the G729 codec (I didn't make the test with G711). The Client configuration is perfect (QoS and bandwidth management). Do you know some issues with the G729 codec? Thanks a lot for your comments, Thomas

Hi all, I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan but I have the following errors and I'm not able to call anymore. Do you know what can I have to do? My Asterisk is connected to a patton with a SIP trunk. [Mar 22 10:19:03] WARNING[16462]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call

...t -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/62a82c69/attachment-0001.htm > > ------------------------------ > > Message: 17 > Date: Wed, 7 Mar 2007 11:17:07 +0100 > From: "Thomas Deillon" <Thomas.Deillon@smart-telecom.ch> > Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > < > 86918CDC1242004D8B056...

...rom: Doug Lytle <support@drdos.info> Subject: Re: [Asterisk-Users] swissvoice To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <42DBD321.2050509@drdos.info> Content-Type: text/plain; charset=ISO-8859-1; format=flowed thomas DEILLON wrote: >Hello, > >I have swissvoice phones and when i use one, a have in asterisk lines like: >Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp >-13691.-232125 > >have a idea ? > > > > Yes, Kevin said this earlier today: 2 wrote: >...

Hello, I think that the best way it's to use vservers: http://deb.riseup.net/vserver/create-instance/ With this you can run as much asterisk as you want :-) Bst regards, Thomas ________________________________ De : asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] De la part de Sri Keerthy Envoy? : lundi, 13. novembre 2006 08:23 ? :

Hi all, I'm trying to send FAX with an anolog fax behind a Patton M-ATA to an other analog fax plug on directly on the PSTN network. I use the last stable version of Asterisk 1.4 ... Somebody have any information why it's doesn't work a all ? Thanks a lot, Thomas

Hi all, I install the Asterisk 1.4.1 in order to use the T.38 pass-through, but for the moment, I cannot even make call .... I have this WARNING: [Mar 7 11:32:09] WARNING[13395]: chan_sip.c:12290 handle_response: Remote host can't match request BYE to call '5759b80c119e1d51679dc66b519c6eac@194.148.41.50'. Giving up. Do you know what is this error and what can I do to

Hi all, Here is my configuration: Phone ?? Asterisk ?? Gateway (SIP 2 PSTN) In the Gateway (patton) I have in "codec order" G729 then G711 If the Phone use G729, I have a pass-thru in the Asterisk ... It's the main case. But If I put G711 in the Phone, I want that the Asterisk try a G711 codec with the Gateway !! because in this case, it's G711 between the Phone and the