search for: schøpzinsky

...e are very seriously considering switching to Sangoma at this moment, due to this and other problems, but I want to know if there is a solution, and to make sure it isn't asterisk that's freezing the cards, and that the problem would re-appear on the sangoma cards. Kind Regards Jon Leren Sch?pzinsky

Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using

...delelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olle E Johansson Sendt: 8. juni 2006 12:09 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Using regcontext 8 jun 2006 kl. 11.57 skrev Jon Sch?pzinsky: > Hello List > > Ive been trying to use regcontext, but I cant get it to work. Ive > setup my sip peers to have the regexten _[0-9]., so that I can > capture all registrations in a single extension. > But when they register, I can see that the dynamic extension is...

Hello List Is there a way to have hints sent between multiple servers? We are currently implementing a cluster solution for our asterisk servers, and the problem is this. User A registers on Asterisk 1 and user B registers on Asterisk 2. User A subscribes to user B's status, through SIP NOTIFY messages. As user B is registered to Asterisk 2, and not Asterisk 1, the NOTIFY messages are only

...----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Kevin P. Fleming Sendt: 9. juni 2006 16:25 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Call status subscriptions on multiple servers ----- Jon Sch?pzinsky <jos@detele.dk> wrote: > Is there a way to replicate subscription info between asterisk > servers? Not at this time, no. That will be probably be worked on during the next development cycle. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. _________________________...

...sending DTMF. Could it be that you are a little bit confused? Usually the problems with DTMF depend on how the phone is configured and how Asterisk is configured (DTMF using SIP INFO, RFC2833 etc), check this out: http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Regards. On 6/14/06, Jon Sch?pzinsky <jos@detele.dk> wrote: > Hello > > How do I get Asterisk to receive DTMF from our Snom phones, when I use G.729? > > Regards > Jon > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.1.394 / Virus Database: 268...

Hello list I've been implementing an asterisk based cluster, and are having grave problems with SIP. My current implementation monitors registrations through the Asterisk Manager interface, but it seems to not register all registrations. Because of this, ive been looking at DUNDi, to implement the cluster. Has anybody done this with success? It seems that it uses the extensions instead of

You can use the CDR(userfield) value, to save the ip's in the CDR record. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Benjamin Jacob Sendt: 3. november 2006 06:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] ip address in CDR Hello ppl, Any way to store

Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Sch?pzinsky

...oming from the same IP as voipsip, but if voipsip is a different user, why on earth does asterisk not identify it as voipsip instead of 12345678??? Some of the values and numbers are changed for security, etc. nonce="xxxx", response="xxxxx" and such. Kind Regards Jon Leren Sch?pzinsky

Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. Jon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Yehavi Bourvine +972-8-9489444 Sent: 19. marts 2007 09:14 To: asterisk-users@lists.digium.com

...: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Philipp Kempgen Sent: 19. marts 2007 09:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Conference server (or how to make a call withmorethan 3 u Jon Sch?pzinsky wrote: > Use Snom phones. > We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. I don't think this is true. The Snoms do not have enough CPU power for 12 people in a conference *on the phone...

...ead of opening a new channel and dialing out. Etc: A calls B on our asterisk, and is directly redirected to C We have been told that this feature should be available on a PRI level, and is called Partial re-routing. Anybody has an idea of whether this is supported in Asterisk? Kind Regards Jon Sch?pzinsky Detele. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.8.9/834 - Release Date: 05-06-2007 14:38

...reating Cell-specific applications, it shouldn't be that big of a challenge to convert the G729 reference code, to Cell. If anybody is interested in trying this, any serious requests can get shell access to the machine, by contacting me off-list. Venlig Hilsen/Kind Regards Jon Leren Sch?pzinsky -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071214/8f27ddc7/attachment.htm

Hello SetCallerPres function seems to be removed from Asterisk 1.4. What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections. Jon No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 -------------- next part -------------- An HTML

Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 - Release Date: 24-10-2006

Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 Til:

...body know of a patch that fixes this problem? I can see that in the end of the T.38 packet, cisco adds 4 zero fields, which are not in the packets that Asterisk sends. Is this some weird "we-are-cisco-and-therefore-decide-how-the-packets-should-look"? Kind Regards Jon Leren Sch?pzinsky Systems Architect Firstcom A/S B?dehavnsgade 2C, 2. 2450 K?benhavn SV Web: http://www.firstcom.dk <http://www.firstcom.dk> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2009...

Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users

...sk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Sch?pzinsky <HYPERLINK "mailto:jos@detele.dk"jos@detele.dk>: Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon What do you mean by...