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...ch is 100 meters further... will a SNOM M9 handheld from base station A register to base station B when it enters its DECT-environment.<br> <br> Can one transparently walk from place A to place B with the same M9 handheld and not loose the conversation ??<br> <br> <br> Greetingz,<br> Jonas.<br> </font></small> </body> </html>

...d this bug (and tried a lot to give me a quick help!). But lowering all Buffers in the mp3decoder.h and something we weren't able to track this bug down. Is anybody else of you involved in the development / knows how to deal with the code? Could someone possibly have a look at it? <p>Greetingz Stefan --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe m...

...to leave a voicemail message. It's to late to answer() the call in the macro, but I guess the voicemail()-application automatically anwers the call ?? How about an IVR-prompt and a queue ? Do I need to answer the incoming call before playing a voiceprompt and before sending it into a queue ?? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/dab3448d/attachment.htm

...on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime sip-database, I could be confronted with NAT-problems for SIP-peers that are behind a NAT-router ? Is this the choice I need to take ? Greetingz, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/2d1523d8/attachment.htm

...pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090413/2a6054c9/attachment.htm

...opnemencallee => #*3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m and I press #*3, nothing happens... No output on the CLI. There's not much info. I followed the instructions on voip-info.org (which are the same as in features.conf). The module res_features is loaded. Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090907/b016a717/attachment.htm

...do this evening. - I have send my IAX-provider an email asking for their iax.conf... - My Asterisk-server is able to contact the remote Asterisk-server but registration is rejected... (REGREJ on CLI). I personally think there is a configuration error in my user-account in their iax.conf-file ... Greetingz, Jonas. >----- Oorspronkelijk bericht ----- >Van : Steve Edwards [mailto:asterisk.org at sedwards.com] >Verzonden : maandag , mei 4, 2009 09:24 AM >Aan : stevend at moij.biz, 'Asterisk Users Mailing List - Non-Commercial Discussion' >Onderwerp : Re: [asterisk-users] Can so...

.../firewall --internet--> Asterisk --internet (back)--> router/firewall (back) --> IP-phone2 So I don't want an Asterisk server in my company (don't have appropriate place) and so I place the Asterisk-server in a datacentre. How about local calls going via the internet and back ?! Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/075dc1f1/attachment.htm

...: Asterisk detects the other end really good and registers when the caller has put down his phone and the conversation is terminated by the caller. Also a fax and a busy-tone is well detected. The option busydetect=yes is set in my chan_dahdi.conf... But this is not the problem. Is this a bug ?? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090512/edbe100c/attachment.htm

...a lot to give me a quick >help!). But lowering all Buffers in the mp3decoder.h and something we >weren't able to track this bug down. > >Is anybody else of you involved in the development / knows how to >deal with the code? Could someone possibly have a look at it? > > >Greetingz > Stefan >--- >8 ---- >List archives: http://www.xiph.org/archives/ >icecast project homepage: http://www.icecast.org/ >To unsubscribe from this list, send a message to 'icecast-request@xiph.org' >containing only the word 'unsubscribe' in the body. No subject...

...skellens-10870", "s-CONGESTION|1") in new stack [May 15 16:55:14] VERBOSE[2743] logger.c: -- Goto (intern,s-CONGESTION,1) What is this circuit-busy ?? What triggers this ?? How can I handle this ?? If the phone was not busy at all... what is then this "circuit" busy ? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090515/f67837d4/attachment.htm

...I must do a 'sip reload' to get registered again. What could be failing ?? Is this a NAT issue of some kind ? Could it be that my firewall at one point blocks things off ??? If it has something to do with NAT or firewall, why does a simple 'sip reload' gets me registered again ?! Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090706/87ec6db3/attachment.htm

...r the 2.6.20-prep kernel installed." I have installed : - kernel-headers-2.6.18-128.4.1.el5.x86_64 - kernel-devel-2.6.18-128.4.1.el5.x86_64 - kernel-xen-devel-2.6.18-128.4.1.el5.x86_64 bash-3.2# uname -r 2.6.20-prep bash-3.2# ls -l /usr/src/redhat/SOURCES total 0 What is the problem ?? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090818/deb0d5e9/attachment.htm

Hi list, quick question : With GoToIfTime, how to define a period of holiday that starts at the end of the month and ends at the beginning of the next month ?? Like September 25th till October 10th when incoming calls need to go to the voicemail... Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090829/d575663c/attachment.htm

...p_addon_sql_mysql.c:335 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'vakantie_data2 FROM AstDB where SIPACCOUNT="092779077"' at line 1 Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090830/8840de37/attachment.htm

...leted the original and replaced it with a "touch vm-intro.gsm". So Asterisk quickly goes on after 'playing' the sound-file and immediately hangs up (which is the next priority to execute). I don't understand why I am not connected with the voicemailbox to leave a message ??? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090917/065a4f0b/attachment.htm

...ormally I work with vi and by editing the config files, but now I've installed the asterisk-gui from svn to check it out. Is there a way to limit access to certain menu's ? Is there a way to create a user with limited access ?? Can I create a user and choose the menu's that one sees ? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091024/f8920079/attachment-0001.htm

...is SIP-member1 transfers the call to another SIP-member2, and this SIPmember-2 rejects the call, then the communication is lost. How can I make the call go back to the SIP-member1 ? Or maybe back to the queue ? To transfer we use the 'transfer'-button on the Grandstream/YeaLink IP-phone. Greetingz. Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100310/a19442c4/attachment.htm

...the executable wine gave a error being unable to find a certain .dll, but it was in the same tmp directory. I thought i copy this .dll to the system directory, didn't work. Anyone have a clue where i go wrong. OS Red Hat 7.2 Wine 20010731 (i know it's old but is that problem?) greetingz Harrie van de Kerkhof

...ocal server detects the 'hungup' @ 17:40:47 and the Asterisk-server in the datacenter @ 17:41:15. There's a lot of time between the moment that I put down the horn of the Grandstream and the moment that my CellPhone stops ringing. Is the delay due to the RTCP SR transmission error ??? Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091003/4cd9488f/attachment.htm