similar to: asterisk sccp support

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

I have 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

Ok so it seg faults when I try to dial out through IAX(voiptalk.org), ofcourse it doesn't if I remove allow=speex :P ---- (gdb) run -c Starting program: /usr/sbin/asterisk -c [Thread debugging using libthread_db enabled] [New Thread 16384 (LWP 28283)] [New Thread 32769 (LWP 28285)] [New Thread 16386 (LWP 28286)] [Thread 16386 (LWP 28286) exited] [New Thread 32771 (LWP 28287)] Asterisk

Hi, Just started to get this error after updating 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

Hello all, I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA) on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one B channel. But when I open a second B chan, the sound is choppy, with too long gaps, and the CPU load is too high (~50%). On the Asterisk's console I get these messages: -- Executing Dial("H323:4478",

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

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal 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

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

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Hi, Just wondering if someone could better explain how the indications.conf file actually affects Asterisk? I am using a Cisco 7940 from my Asterisk system, and have set in indications.conf "country=au" thinking that this would make the dialtones/call progress sound like the familiar Australian tones? However when I call another extension on my system, it still sounds like

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!!!

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