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Thanks in advance root@asterisk01:~# modprobe wcfxo /lib/modules/2.4.20/misc/wcfxo.o: init_module: No such device /lib/modules/2.4.20/misc/wcfxo.o: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/mod...

After working fine for a week or so my new Quad E1 asterisk 1.8 system has started rejecting outbound calls from the Nortel BMC 450 it is connected to. The cli fills up with these: sig_pri.c: Ring requested on unconfigured channel 255/255 span 3 Is this likely to be a 1) config error 2) cable issue (I made them) 3) hardware problem with the Digium card 4) software (lib pri) Any clues?

...rver running latest asterisk 1.4.x connected to a Siemens hi-path user a TE120P single-span. Approx every 8 hours (although not every 8 hours and sometimes 2 in a row) at exactly the same time I see the following errors Does anyone have any suggestions / ideas ? Thanks Ruairi Sep 24 11:56:30 asterisk01 -- MARK -- Sep 24 11:59:02 asterisk01 ntpd[3615]: synchronized to 172.16.22.60, stratum 4 Sep 24 11:59:28 asterisk01 kernel: wcte12xp: NMF workaround on! Sep 24 11:59:28 asterisk01 kernel: wcte12xp: Setting yellow alarm Sep 24 11:59:29 asterisk01 kernel: wcte12xp: NMF workaround off! Sep 24 11:59:3...

...sion" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by 'pbx_config' ] '0001' => -1. Dial() [pbx_config] 1. Dial(SIP/mvickers) [pbx_config] '0002' => -1. Dial() [pbx_config] 1. Dial(SIP/jgb) [pbx_config] '0003...

...I just arrived in Italy from Cebit, qhere I spoke with digium and Beronet people. They told me to try to use the mISDN stack to drive beronet and the new upcoming digium ISDN Cards. SO I searched, find http://www.beronet.com/download/card_installation_guide.pdf, and I immediately got the error: asterisk01:~ # cd /usr/src/install-misdn/ asterisk01:/usr/src/install-misdn # make install CONFIG_SMP=y !!!!!!!!!!!!!!!!!!!!!!!!!! Disable the SMP Setting in your Kernel Config. make: *** [test_preempt] Error 1 So I discovered that mISDN does not support SMP and preempitive multitasking. but how can I...

...rce.com Subject: [Xen-users] Missed interrupt. Increasing latency to 5 ms in orderto compensate. I am using pciback to pass a Digium PCIX analog telephony card to an Asterisk domU. Seemingly at random times, the card wigs out with the following appended to my /var/log/messages: Jun 10 12:17:19 asterisk01 kernel: wctdm24xxp0: Missed interrupt. Increasing latency to 4 ms in order to compensate. Jun 10 14:01:01 asterisk01 kernel: wctdm24xxp0: Missed interrupt. Increasing latency to 5 ms in order to compensate. This is becoming a major problem because it will drop any active calls and refuse to accep...

...apter to my linux box (Suse 10) i see: Bus 005 Device 001: ID 0000:0000 Bus 004 Device 003: ID 06e6:831c Tiger Jet Network, Inc. Bus 004 Device 001: ID 0000:0000 Bus 003 Device 001: ID 0000:0000 Bus 002 Device 001: ID 0000:0000 Bus 001 Device 001: ID 0000:0000 and in warning log: Feb 22 15:46:12 asterisk01 kernel: zaptel: module not supported by Novell, setting U taint flag. Feb 22 15:46:12 asterisk01 kernel: wcusb: module not supported by Novell, setting U taint flag. Feb 22 15:46:12 asterisk01 kernel: Wildcard USB FXS Interface driver registered So everything seems to be OK. Then I modify to the...

...span is configured and kernel module is loaded. If configured correctly and a circuit or channel bank is connected the LED should turn GREEN. ... Below are the output of dmesg, lsmod, /etc/zaptel.conf and ztcfg -vvv. Did anyone have this problem before? Thanks Ant?nio The?philo J?nior root@paggo-asterisk01:~# dmesg | more ... [42949385.790000] Zapata Telephony Interface Registered on major 196 [42949385.790000] Zaptel Version: 1.2.10 Echo Canceller: KB1 [42949385.840000] e100: Intel(R) PRO/100 Network Driver, 3.4.14-k4-NAPI [42949385.840000] e100: Copyright(c) 1999-2005 Intel Corporation [42949385.96...

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16...

...57:13 DEBUG[25689] pbx.c: Function result is '(null)' Feb 9 10:57:13 DEBUG[25689] pbx.c: Function result is 'asterisk-25591-1139479028.6' Feb 9 10:57:13 DEBUG[25689] pbx.c: Function result is '(null)' I tried several configuration, all the same problem. here are my files: asterisk01:/etc/asterisk # cat /etc/zaptel.conf # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 asterisk01:/etc/asterisk # cat /etc/asterisk/zapata.conf...

Hi, I have a connection between a te120p and a siemens hipath which was recently set up. The connection uses etsi / e1 with no crc. Every few hours the asterisk shows the pri as in alarm asterisk01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN and I have to run ztcfg to clear the alarms asterisk01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: Network Window Length: 0/...

...s converted to SIP, and a cisco 2600 xl w/ E1 and SIP. I?m thinking to expand the test adding more 7912, but I prefer not convert all the 7912 to SIP, so I?m tying to put CHAN_SCCP to work. I?ve get the sources from cvs.sourceforge.net but when I try to compile, get the followings errors: [root@asterisk01 chan_sccp]# make Now compiling .... chan_sccp.c 695 lines In file included from chan_sccp.c:15: /usr/src/asterisk/include/asterisk/utils.h:22: redefinition of `struct ast_hostent' chan_sccp.c:50: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC...

Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten => 202,1,ANSWER() exten => 202,2,PLAYBACK(tt-monkeys) exten => 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [202 at local:1]

.... The first VM was ESXi (Free version). The second VM was again ESXi (but the licensed version). We used the VMWare converter to migrate from the Free to the Licensed environment. After we moved Asterisk from the first to the second VM and tried starting it, we got the following message: root at Asterisk01:~# service asterisk start * Starting Asterisk PBX: asterisk Illegal instruction (core dumped) I recompiled Asterisk and tried again. Now I just get: root at Asterisk01:~# service asterisk start * Starting Asterisk PBX: asterisk But during the compiling of asterisk I got these messages: ../src/pj/...

Hello, I'm trying to get the following situation. Someone calls an application on one of our asterisk server. In this application the caller will call a SIP client. (with the command Dial) The Sip client is connected with another asterisk server. (see below) Caller --> asterisk01 (incoming server) --> asterisk00 (outbound server) --> SIP client (X-lite) Do anybody now how the command is to call a SIP client on another asterisk server? It is possible to send the configuration file, just say which configuration files (sip.conf, etc) First I tried to a simpler s...

...ly secure. In addition, blocked IP addresses are shared via the service so you can pre-emptively block addresses that others experienced attacks from. We offer a FREE one week no obligation trial, and during the month of January we also offer a DISCOUNT for Asterisk users. Just enter voucher "asterisk01" at the checkout to receive 30% off. Protection is available from just US$84 per server with the discount. You can read more and give the free trial a go at: http://easysysadmin.com/ Our standard security package includes: - Monitor VoIP traffic (SIP registrations) and block attackers....

...in new stack -- Executing Dial("SIP/567-bb09", "misdn/1/3481303063") in new stack -- Called 1/3481303063 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) I am using Suse Linux 10, and I switched to default kernel (not SMP) asterisk@asterisk01:~> uname -r 2.6.13-15.8-default Any help will be gratly appreciated. by the way: I read it could be possible to use chan_capi insted of chan_misdn, laying on misdn: is it correct: ? And if it is, could anybody give me an advice on how ? I tried the 0.6.4 chan_capi version I succesfully instal...

...ting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel => 1-15 channel => 17-31 asterisk01*CLI> zap show channel 1 Channel: 1*CLI> File Descriptor: 14 Span: 1k01*CLI> Extension: CLI> Dialing: noCLI> Context: default Caller ID string: Destroy: 0*CLI> InAlarm: 0*CLI> Signalling Type: PRI Signalling Owner: <None>I> Real: <None>LI> Callwait...

Hi, after I fixed my problem with the playback() application, I now have the next strange one. When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If some in the town of this client calls him, he gets the ISDN error "service not available". Out office is connected to he office

..."two" "three" perfectly. >> >> >> What do I need to do to hear the extension where the call gets parked? >> > > > > ------------------------------ > > Message: 18 > Date: Tue, 15 Jan 2008 08:17:19 +0100 > From: Stefan Guenther <asterisk01 at in-put.de> > Subject: [asterisk-users] pickupchan without bristuffed version? > To: asterisk-users at lists.digium.com > Message-ID: <478C5DFF.7090008 at in-put.de> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > Hello, > > following the descri...