search for: 262161

...just rebuilt my * box back to last weeks 06-20 CVS build beacuse after getting the latest I could not hear ANY voice prompts. I have a T1 card and a dual proc box that has been running just fine up till this weekend. I tihnk some of the format changes affected my install. Jun 27 16:12:38 DEBUG[262161]: File chan_sip.c, Line 612 (create_addr): Setting NAT on RTP to 0 Jun 27 16:12:38 DEBUG[262161]: File chan_sip.c, Line 523 (__sip_ack): Stopping retransmission on '7ceb145123fa12fc73729d134d2820d8@155.97.244.130' of Request 102: Found Jun 27 16:13:18 DEBUG[262161]: File chan_sip.c, Line 34...

...dating to the latest CVS. Asterisk dies if it can't create a channel - not so good. -- Executing SetCallerID("SIP/750-2550", "39660426") in new stack -- Executing Dial("SIP/750-2550", "CAPI/39660426:22179808") in new stack Jun 22 13:52:05 NOTICE[262161]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 39660426. you should check your config! Jun 22 13:52:05 NOTICE[262161]: app_dial.c:681 dial_exec: Unable to create channel of type 'CAPI' Killed [root@pbx root# Ouch ... error while writing audio data: : Broken...

.........................Sep 30 10:51:12 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call 333ab105721da3172443a8582d1d5ae9@192.168.0.201 for seqno 102 (Non-critical Request) ................................ ] Asterisk Ready. [Thread 245776 (LWP 28302) exited] [New Thread 262161 (LWP 28305)] Sep 30 10:53:40 WARNING[262161]: codec_speex.c:166 speextolin_framein: Out of buffer space Sep 30 10:53:40 WARNING[262161]: codec_speex.c:166 speextolin_framein: Out of buffer space Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 262161 (LWP 28305)] iax2_se...

...pi: appl 1 ncci 0x20201 up -- CAPI[contr1/7810] answered H323:4478 ERROR[344086]: File chan_capi.c, Line 900 (capi_write): error sending DATA_B3_REQ (error=0x1103, datalen=160) ERROR[344086]: File chan_capi.c, Line 900 (capi_write): error sending DATA_B3_REQ (error=0x1103, datalen=160) ERROR[262161]: File chan_capi.c, Line 900 (capi_write): error sending DATA_B3_REQ (error=0x1103, datalen=160) ERROR[344086]: File chan_capi.c, Line 900 (capi_write): error sending DATA_B3_REQ (error=0x1103, datalen=160) ERROR[262161]: File chan_capi.c, Line 900 (capi_write): error sending DATA_B3_REQ (error=...

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... and looking in the code I'm assuming this is bad :-) If it matters AT&T broadband is my phone service and its coming through cabl...

...esResMessage Device has 6 Capabilities -- CODEC: 4 - G.711 u-law 64k -- CODEC: 2 - G.711 A-law 64k -- CODEC: 11 - G.729 -- CODEC: 12 - G.729 Annex A -- CODEC: 15 - G.729 Annex B -- CODEC: 16 - G.729 Annex A+Annex B == >> Got message ButtonTemplateReqMessage WARNING[262161]: File sccp_actions.c, Line 144 (sccp_handle_button_template_req): Don't have a button layout, sending blank template. == Sending Packet Type ButtonTemplateMessage (100 bytes) == >> Got message SoftKeyTemplateReqMessage == Sending Packet Type SoftKeyTemplateResMessage (656 bytes)...

Hi to all!! I'm successful to connect Asterisk to MySQL database... Can anyone learn me how to store sip user in MySQL database and how to configure voicemail?? Thanks for all!!!

...inimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am pretty convinced the SIP setup is OK. This is the error message: Jan 29 12:21:54 NOTICE[262161]: app_dial.c:527 dial_exec: Unable to create channel of type 'IAX2' when i try to call the PSTN from the SIP device. i've tried the wiki, the handbook, the voicepulse site, and all sorts of other sites, and nothing helps. i also downloaded and compiled the code today (jan 29) and tha...

...au' and at the end proceeds to set default indication country to 'au'. the Removed part has me thinking it's forgotten all about the particular indications for au? ======cut from Asterisk console======= -- Unregistered indication country 'us' Jan 18 14:02:36 NOTICE[262161]: indications.c:390 ast_unregister_indication_coun try: Removed default indication country 'au' -- Unregistered indication country 'au' -- Unregistered indication country 'fr' -- Unregistered indication country 'de' -- Unregistered indication cou...

I've been happily using R on SunOS/Solaris for a while, but am now trying to develop some course work on Windows. I know the real action is on Win32, but I have access to a laptop which runs 3.1 and I thought it might be useful to develop on the lowest common denominator anyway. I downloaded "tmp.zip" (dated 03 September 1998) from /R/CRAN/bin/windows/windows, which unzipped to

Hi, I have a new phone in our IP phone network: Planet VIP-101T. When calling from that Planet phone to anybody, everthing is fine. But when calling from anybody to that Planet phone, I get a mashine gun noise and the following msg in asterisk log: NOTICE[262161]: File channel.c, Line 1091 (ast_read): Dropping incompatible voice frame on H323:0 of format SLINR since our native format has changed to ULAW Both, the Planet phone and the asterisk oh323 channel, have G.711A as preferred codecs. For me, it seems, that the planet phone does not follow tha...

...itter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f", "http://stream.lrz-muenchen.de:31337/m945-lq.mp3") in new stack Aug 2 18:16:16 DEBUG[262161]: chan_sip.c:817 __sip_ack: Stopping retransmission on '7e3ee6f44f3fabf0@*' of Response 27022: Found Aug 2 18:16:19 NOTICE[475156]: app_mp3.c:91 timed_read: Selected timed out/errored out with 0 Aug 2 18:16:19 DEBUG[475156]: app_mp3.c:175 mp3_exec: No more mp3 Playing back files from a...