similar to: yet another snom issue

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/949c1368/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031215/9327b656/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try

The following error messages were observed in /var/log/asterisk/messages: Sep 22 10:26:42 NOTICE[1133735216]: File chan_sip.c, Line 5099 (handle_request): Unknown SIP command 'PUBLISH' from '212.23.220.236' Sep 22 11:32:50 WARNING[1133735216]: File chan_sip.c, Line 4519 (handle_response ): Got 200 OK on REGISTER that isn't a register The phone was a Snom 200 running v2.01t

Hi List I'm getting this notification from my one and only SNOM 360 every time a number button is pushed. I know that it's only a notification, but it really irritates me. Is it anything I can/should do anything about ?? Oct 12 10:34:33 NOTICE[3566]: chan_sip.c:10530 handle_request: Unknown SIP command 'PUBLISH' from '192.168.100.100' By the way I'm

Hi all, Anybody know how to configure Asterisk (sip.conf and extension.conf) to work with an Adtech Tipcome SIP Phone s-150? Using a conf like this: exten => 100,1,Dial,SIP/100@adtech (in extension.conf) and [adtech] type=friend username=123 secret=123 host=dynamic defaultip=192.168.0.10 (in sip.conf) I have "Registering Error!" on display and NOTICE[5126]: File chan_sip.c,

Hi all, I have been struggling desperately to get * work together with my snom100 for days on end, but I am not making any progress... Of the entries marked *#) I'm still not sure what it does; so far I have on the snom in "SIP/lines" -user name - empty *1) -account - Conrad -registrar - 192.168.200.83 -action - "None" *2) in

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

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

(Dovecot 1.0.rc15) Hello I'm tring to configure postfix+dovecot on debian etch. In dovecot I use passdb pam and userdb passwd-file. I can connect pop3 clients and reveice mails and connect smtp clients and send mails, but it is delivered to /var/mail/%u instead the configured mailboxes in /var/local/dovecot/%u. I`ve configured postfix as say http://wiki.dovecot.org/LDA/Postfix.

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

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 left these information out sorry = Detected 1 licensed G.729 transcoders WARNING[1024]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 -------------- next part -------------- An HTML

Hello. If I receive a Phone call by ISDN or from SIP Provider, the Asterisk make some errors and the SIP Client don't react. The messages from Asterisk in verbose mode: er will net. Asterisk messages in Terminalmode: parse_srv: SRV mapped to host sip-ha.web.de, port 5060 Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown"

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

hi, i just ported the patch of David Hinkle http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html to the current cvs version of asterisk. the theory is that the leds of supervisioned extensions are blinking until a call is established whereafter the leds should be constantly lit. however it's not working. the asterisk server is sending the following xml notify to the

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 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

I too am having this problem reported by Frank Hoonhout. Asterisk runs fine for a few minutes and then stops accepting new calls. (I have a standalone server with SIP phones and I'm not doing any external registration). Asterisk CVS-04/07/03-09:28:50 0x420e0037 in poll () from /lib/i686/libc.so.6 (gdb) info threads 16 Thread 14351 (LWP 7258) 0x420e187e in select () from