similar to: SVN Server Issue?

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.18 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.18 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

I have been using Asterisk 1.0-RC2 successfully with a channelized T1 circuit for quite a while now but after upgrading to 1.0.0 callerid no longer works properly. Debug output from a channel shows what actually is received through DTMF from the carrier: << [ TYPE: DTMF (1) SUBCLASS: * (42) ] [Zap/9-1] << [ TYPE: DTMF (1) SUBCLASS: 2 (50) ] [Zap/9-1] << [ TYPE: DTMF (1)

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a sample of the issues resolved in this release: *

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a sample of the issues resolved in this release: *

Thanks Bruce, and elaborately as ever. Again, I'm surprised about your very thorough Z80 knowledge when you said you only did little on the ZX81 in the eighties :D OK, understood. I was first thinking about doing something like this for small frames: 1. push bc # 1 byte; 11 cycles - part of call frame-cleanup: save scratch register +-----begin call-related 2. ld

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified field. ""-any custom value that you wish to store."" My question is how do you setup more custom fields in the cdr and be

>It should be possible to get llvm to produce very good code for the Z80... Yes, I was thinking that too. These techniques didn't exist back then, so I'm really looking forward to the point where the first regular C sources can be compiled and see the magic happening in action live :) ------------------------------------------------------------------------ *From:* Bruce Hoult

Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Tx-Frame Retry[000] -- OSeqno:

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

Hello everyone, I would like to seek your advice regarding a build (or rather "configure") problem I am running into. For reference, tests are all relative to a build from a 1.8.26.0 tarball, on Debian Wheezy. I would like to understand if it is possible, and if any of you have tried, to build Asterisk from a subdirectory of the source tree, i.e., from a clean source tree # mkdir

Hi, In /var/log/asterisk/full (Asterisk 1.6.2-rc6), I can see : [Dec 8 15:02:17] VERBOSE[10283] config.c: == Parsing '/var/spool/asterisk/voicemail/default/103/INBOX/msg0000.txt': [Dec 8 15:02:17] VERBOSE[10283] config.c: == Found [Dec 8 15:02:17] VERBOSE[10283] file.c: -- <SIP/103-00000784> Playing 'vm-message.gsm' (language 'fr') [Dec 8 15:02:18]

As discussed previously: https://www.redhat.com/archives/libguestfs/2010-July/msg00003.html https://www.redhat.com/archives/libguestfs/2010-July/msg00024.html To do: (1) Implement progress notifications for many more daemon operations. (2) OCaml bindings to the callback. (3) Perl bindings to the callback. Rich. -- Richard Jones, Virtualization Group, Red Hat

For a Z80 backend, "eliminateCallFramePseudoInstr()" shall adjust the stack pointer in three possible ways, e.g. after a function call, depending on the amount (= adjustment size) *and some other rules*: 1. via one or more target "pop <reg>" instructions (SP increments +2 per instruction), using an unused reg (disregarding the contents after the operation), followed

Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under

The Asterisk.org development team is proud to announce the release of Asterisk 1.4.23. This release is available for download from http://downloads.digium.com/. This release is a significant bug fix update for the 1.4 release series. The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help! * Fixed