similar to: 11.13.1: unable to load sip.conf (or iax )

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Asterisk don't running, because show this message WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' lacks type WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing interface: No such file or directory WARNING[6949]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled WARNING[6949]: chan_oss.c:239 sound_thread: Read

Dear Asterisk developers, I run the same asterisk version on the home machine and on the work. On the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work machine I have Mandrake 10.1 (kernel 2.6.8.1 <http://2.6.8.1>). When I run asterisk on the work machine, these warnings and error appear (there are no warnings or error at home): [ Booting......Oct 17 18:19:04

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

All, I signed up for a broadvoice BYOD plan over the weekend (very excited about their offering) and after about an hour I had asterisk registered and was making in and out bound calls. However, the next day (without changing anything) I couldn't call in or out and haven't been able to get it going again. I can connect using a softphone (X-Lite) and make calls in and out

I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using

Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Is there some method to use a Windows program icon in Wine? droid

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

Hello, Can't get chan_gtalk.so module to load, neither res_jabber.so: Asterisk*CLI> module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory [Dec

I'm trying to set up csipsimple on my Droid X. But no joy. Can't get it to register. My sip.conf: [general] ........ tcpenable=yes ........ [Test] transport=tcp,udp type=friend secret=mytest host=dynamic context=cloud-out qualify=60 dtmfmode=auto insecure=port,invite disallow=all allow=ulaw I've tried both udp and tcp. Does anyone have csipsimple working? Could you share your

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

Hello, i have a simple but annoying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten =>

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

Hello, I'm looking to use tinc. What are my options for connecting from a mobile device, e.g. iPhone/iPad or a droid? Is there any backwards compatibility with IPSec or PPTP or anything? Or would it be possible to implement a solution where a pptp vpn is bridged to the tinc vpn so clients can 'see' other clients on either network? Thanks, Adam

Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 -

I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (39) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host