similar to: Hmmm - anyone seen this before?

I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER <sip:714XXXXXXX@1.0.24.5>;tag=7f8072c0c46250f7o1 X's are there to

Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious.

Can Asterisk light the message waiting light on a Grandstream BudgeTone phone? If so please reply with any related configlets :-) Thanks!

Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks.

I recently swapped 2 FXO modules on to what had previously been a 4 FXS version of the TDM400 board. The FXS ports are recognized - the FXO ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both say channel 1 isn't there). Has anyone experienced this problem? All software is current as of this AM. If the old FXS modules are re-installed all works just dandy (other than

Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '117801284512845hUxv-9991110061--17708185305@63.201.117.76' of Response 20587: Found Jul 6 15:12:10

Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

In my code here https://github.com/DragonOsman/currency_converter , I used C++17 and managed to get it to work (though I'm only using std::map::insert_or_assign() from C++17). And I'm using Windows, so I shouldn't use LDFLAGS or CXXFLAGS as environment variables. I'll use them directly on the compiler command line instead. The libraries I need to link against are

Are you linking with a C++ compiler? A lot of those missing symbols look like they come from the C++ standard library. -David Osman Zakir via llvm-dev <llvm-dev at lists.llvm.org> writes: > @blubee blubeeme So what do you think? Got any ideas? > ---------------------------------------------------------------------- > From: Osman Zakir <osmanzakir90

I had everything working fine until today. Today the Sipura cannot dial anywhere. I just get the following: Feb 10 12:48:18 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:19 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:35 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

I couldn't get it to build libcxx... You need c++ and c++abi to compile c++ code. On Wed, Dec 12, 2018, 07:01 Osman Zakir via llvm-dev < llvm-dev at lists.llvm.org> wrote: > LLVM on a Developer Command Prompt. The ones I want to fix first are the > ones from Boost and Jinja2Cpp. I saw some from those as well. > > If there any standard library ones missing, could it be

So how do I get it to build libcxx and libcxxabi? I got it from the mono repo and enabled lld, clang, libcxx and libcxxabi. But I built the two main CMake targets only--all_build and install. What else do I have to do? Please let me know. ________________________________ From: Zachary Turner <zturner at google.com> Sent: Wednesday, December 12, 2018 11:10 AM To: blubee blubeeme Cc: Osman

How can I tell CMake during the configuration step where to find my zlib installation? ________________________________ From: blubee blubeeme <gurenchan at gmail.com> Sent: Wednesday, December 12, 2018 7:31 PM To: Osman Zakir Cc: llvm-dev Subject: Re: [llvm-dev] Using LLD to link against third-party libraries? How? I would agree with the next email from Brian Cain If you do not have

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

I already mentioned what flag I tried. It's in the first email in this thread. And I want to link against Boost.System and the Jinja2Cpp library (the latter's documentation can be found here: https://github.com/flexferrum/Jinja2Cpp . And I also have some GUI applications using FLTK as well that I want to try to build using LLVM as well, so I'll have to know how to link against

Hi, I have installed chan_sip on asterisk-1.0.3 / 5 (tried both, same result). My sip phone registers fine. But when dialing a number, I get: Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to create/find channel ... Feb 2 09:44:52 WARNING[20380]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call 384534305@192.168.1.20 for seqno 219 (Non-critical Response)

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

I have just set up my first Asterisk, and I have the basics up an running. I have set it up with two SIP phones only. I can call between them, and I can call out to FWD phones. However, on receiving calls from FWD, my Asterisk blocks the calls with the following message: May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_request: Failed to authenticate user <user>