search for: wemheuer

Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 => , AB,,,tz=cet|attach=no| Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an

Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi

Karsten Wemheuer <kwem at gmx.de> schrieb: Hi Karsten! > the timeout value of 15 minutes directs me to an issue with session > timer. Try to refuse them by putting the line > session-timers = refuse > into the general context of sip.conf. Reload the sip stack with "sip > reload&q...

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem

Am 17.06.2020 14:37, schrieb Karsten Wemheuer: Hi Karsten! > The product is "All-IP" and not the SIP trunk, right? > The call starts normally and after about 15 minutes the quality is > disturbed? No, current we have Magenta Zuhause. Tomorrow we'll change to DeutschlandLAN IP (business contract). The quality is distu...

...ase of Asterisk 10.1.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix ACK routing for non-2xx responses. (Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer) * --- Fix regressions with regards to route-set creation on early dialogs --- (Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.3 Thank you for...

Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but

...dia path. Is this a new bug or do I something wrong? File sip.conf looks like this: [general] bindaddr = 0.0.0.0 disallow = all allow = alaw allow = ulaw language = de allowguest = no fromdomain = 192.168.10.70 tos_sip = 96 tos_audio = 184 [katrin] type = friend host = dynamic callerid = Katrin Wemheuer <200> context = Standard mailbox = 200 [max] type = friend host = dynamic callerid = Max M?ller <245> context = Standard mailbox = 245 Thanks, Karsten

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

Hi, I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel panics on a very small load. The first call normally gets through. If I start the second or third call

Hi Karsten, On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > .... The server sends Server Hello, Certificate, Server Key > Exchange and Server Hello Done. Something in that packet seems to be unacceptable for openssl 1.1.1d as it is compiled and configured for Buster. Certificate length, Digest algorithm, ... You my change the syste...

...elease: ----------------------------------- * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis...

...elease: ----------------------------------- * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis...

...s abandoned (Reported by Kfir Itzhak) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ram...

...s abandoned (Reported by Kfir Itzhak) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ram...

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten

Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature "partial rerouting" which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user

...elease: ----------------------------------- * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis...