Re: [asterisk-users] [OT?] Elmeg IP290: do someone know this telephone?

Hi, It's a branded Snom 190... Thanks, Carsten Luca Bertoncello schrieb am So., 30. Aug. 2020, 13:35: > Hi! > > I have a little problem with the given phone... > Do someone know it? My problem is that I'd like to display the name of > the caller (if it is saved in the address book, of

Re: [asterisk-users] What does pct mean?

Hi, Lost percent (%) Thanks, Carsten Am 11.02.2018 19:27 schrieb "Michael Maier" : > Hello, > > could somebody please tell me the meaning of "Pct" as seen in asterisk cli: > > ...Receive. .Transmit.. > CountLost Pct Jitter

[asterisk-users] SIP-Header Pass-Through

release from GIT. If I can confirm it with the latest 13.x from GIT, should I post my findings to the dev-list or is there some security mailing list, as this would apply to all installations around there? -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germ

[asterisk-users] 1.4.21.2: Linking res_crypto causes segmentation fault.

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: -- [CC] res_adsi.c - res_adsi.o [LD] res_adsi.o - res_adsi.so [CC] res_agi.c - res_agi.o

Re: [asterisk-users] 1.4.21.2: Linking res_crypto causes segmentation fault.

: On Wed, Jul 23, 2008 at 10:12:21AM +0200, Carsten Bock wrote: Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: -- [CC] res_adsi.c - res_adsi.o

Re: [asterisk-users] 1.4.21.2: Linking res_crypto causes segmentation fault.

I just re-unpacked asterisk-1.4.21.2.tar.gz and there was no '`' in the function mentioned below. I have no idea where it came from, i didn't edited the file before. May be something is (terrible) wrong with the server i installed it on ... :) = now it compiles and works Carsten Bock schrieb

Re: [asterisk-users] chan_features.so / asterisk 1.4.5

José Luis Ledesma schrieb: In my asterisk 1.4.5 chan_features.so has been installed properly... check in your asterisk-source if /channels/chan_features.so is present regards, Jack escribió: Is chan_features.so deprecated for asterisk 1.4.5 or why is this module not installed by

Re: [asterisk-users] snom led not working with asterisk 1.4.1

Jamie Heckford schrieb: Just to let you know that this doesn't work with the latest SNOM firmware. With the latest FW 6.5.8 the LED work in most cases (for Unavailable , see below) on my SNOMS. BUT: They don't show the correct incoming account/line: only the LED for the first accounts lights

Re: [asterisk-users] snom led not working with asterisk 1.4.1

Steve Murphy wrote: On Wed, 2007-03-21 at 15:00 +0100, Giorgio Incantalupo wrote: Hi Steve, as you know if you type show hints inside asterisk console you can see phone status. When a phone is not connected, Asterisk says it is Unavailable. With Asterisk 1.2.9.1 my SNOM leds worked well so

[Asterisk-Users] G726-40 required - Help!

Hi everybody, A customer requires G726-40 with Asterisk... I know G726-32 is pseudo-standard, but he definitely wants G726-40... Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone done this before? Any hints? Please help! Due to a misunderstanding, my product manager

[Asterisk-Users] G726-40 required - Help!

Hi everybody, A customer requires G726-40 with Asterisk... I know G726-32 is pseudo-standard, but he definitely wants G726-40... Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone done this before? Any hints? Please help! Due to a misunderstanding, my product manager already

[Asterisk-Users] agi debug - unable to set normal priority

In my agi-debug i get the following error-message: AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: I have the same problem with all (shell) AGIs. Not sure when it started (about two days ago) and why, i tried to restart asterisk and my server and also reinstalling

[Asterisk-Users] agi debug - unable to set normal priority

Carsten Bock wrote: In my agi-debug i get the following error-message: AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Oups there something missing, the complete error message is AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Unable to set

[Asterisk-Users] Problems with a SMS-capable Phone on a ZAP Channel / Question about native bridging on digium cards

Hi Everybody, We just changed our setup from a classic PBX connected directly via PRI to the PSTN to a PBX-Asterisk-PRI-PSTN Setup. My question: Does anyone on this list know, how the Digium Cards handle the native bridges between two Zap-Channels? Is it handled on the Digium card or by the

[Asterisk-Users] Noises on ZAP Channels

a hint what to do/improve or what to search for. So far i did not find anything useful on the net Thanks in advance, Carsten Bock ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Creating CDR's with online connected time

Hi, el Flynn wrote: Carsten Bock wrote: Hi there, How do i setup asterisk, so that in the CDR's is only the time, which the line actually was connected? Not the time, the line was up, but the time the user was able to talk to another user. isn't it already in there? I guess not. Here

Re: [Asterisk-Users] Creating CDR's with online connected time

Hi Eric, Thanks for the hint. Now i know, what i have to search for; and as far as i can see, this will bring me to a solution... Carsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] Problems with not correctly unregistered users...

, who have the same IP/port. If there are no solutions/experiences on this list, i will send my message to the -dev list as well. Thanks in advance, Carsten Bock ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Creating CDR's with online connected time

Hi there, How do i setup asterisk, so that in the CDR's is only the time, which the line actually was connected? Not the time, the line was up, but the time the user was able to talk to another user. Thanks in advance, Carsten ___ Asterisk-Users