search for: sikkema

...quot;Wildcard TDM400P REV E/F Board 1" in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it up for timing only. What do I have to do (I have no experience at all with zap channels and the zaptel.conf file)? -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

...ace, I'm sorry for that). I've tried to read chan_sip, but couldn't find something in the time I had today. I've found hardly any documentation o this variable, apart from that it exists and that it contains "the" SIP CallID value. Can anyone enlighten me? -- Andreas Sikkema -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060616/f49a559f/attachment.htm

I've successfully got inbound/outbound calling working with our Asterisk using the Asterisk-OH323 channel driver. We are using a parent gatekeeper and the NuFone H323 channel driver would not work with the parent gatekeeper... I'm trying to determine a way to ensure that the line used for outbound calling is always available i.e. like trunking.. >From what I can tell when I place an

...5 WARNING[467828746]: RTP Read too short Sep 9 09:25:45 WARNING[467828746]: Asked to transmit frame type 256, while native formats is 32 (read/write = 256/256) With lots of the last variety. Does Asterisk support T.38 (froma Google search it seems not), is anyone working on this? -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

...t; Asterisk We've seen a problem here with asterisk. Wehn Asterisk sends it reinvite, it uses its own codecs, not those of the other endpoint. So until someone fixes that (when possible), there's no way this will work. We're using a CVS version of approx. a month ago. -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: >> but as soon as I configure another sip registration on another server, >> outgoing >> calls drop after 32 seconds. > Are both your servers behind the same NAT router? > thanks for taking part... I don?t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr...

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:

Hi, all!!! What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case? All phones there in LAN behind Asterisk prefer GSM codec, so it does transcoding. So, what I mean is will Asterisk fall back to use

...;Wizard 'config'" to be applied or why I need it. Googling for this phrase suggested I created an empty config file for pjproject but this also didn't resolve this problem. I am sure I must have missed something, can someone point me in the correct direction? Thanks! -- Andreas Sikkema

...unately Asterisk doesn't seem to understand what I want it to do: Aug 6 15:46:42 NOTICE[524301]: pbx.c:4700 pbx_builtin_gotoif: Not taking any branch I've been looking all over the wiki and google, but can't find any example doing what I want to do. Is it even possible? -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Hi there, I'm using ser and asterisk together. Asterisk for voice mail etc and ser for registration of the users usig database. I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Thanks in advance, Pavel -------------- next part

Hello, I would like to use Intel Blade machine for running Asterisk. Is there anyone who already use Intel Blade server for running Asterisk? Can you please explain, how perform Asterisk with Intel Blade machine? I would appreciate for giving me feedback regarding this issue. Regards Nahid -------------- next part -------------- An HTML attachment was scrubbed... URL:

[thread moved from -dev due to non-dev content] At 6:40 PM +0200 on 8/13/05, Andreas Sikkema wrote: >On Sat, 2005-08-13 at 12:44 +0800, Steve Underwood wrote: > >> He doesn't seem to really understand the strengths and weaknesses of >> IAX. IAX has drawbacks, but none of the problems he lists actually exist. > >OK, I'll bite ;-) > >How would IAX2 sol...

...en when a SIP "ATA" 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

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

...[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yves A. > Sent: Saturday, November 22, 2014 8:06 AM > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when.... > > Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: >>> but as soon as I configure another sip registration on another server, >>> outgoing >>> calls drop after 32 seconds. >> Are both your servers behind the same NAT router? >> > thanks for taking part... > > I don?t know... > one is > > si...

Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

...gt; asterisk-users-bounces at lists.digium.com] On Behalf Of Yves A. > Sent: Saturday, November 22, 2014 8:06 AM > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only > when.... > > Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: > >> but as soon as I configure another sip registration on another server, > >> outgoing > >> calls drop after 32 seconds. > > Are both your servers behind the same NAT router? > > > thanks for taking part... > > I don?t know... > one is > &g...