similar to: WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type

Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --

Hi, I live in Italy and I have an Asterisk 1.0.7 box with a HFC monoBRI card connected to my cisco router which is connected to FastWeb provider: does anybody knows why every time my cisco router turns off, my telephone connection to Fastweb drops (while internet connectior is ok)? Restarting Asterisk is worth nothing. TIA Giorgio --

There is no static interval. But i found out that it was my IP-Phone Service Provider that was having serviceproblems today. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Giorgio Incantalupo Sendt: 26. august 2005 11:33 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users]

Hi, I've upgraded my Asterisk to 1.0.7version patched with bristuff 0.2.0-RC8c. I'm using the same extensions.conf but it seems now include instruction doesn't want to work, here follows an extract: [inbound_menu] include => ins_exts exten => _X.,1,Answer exten => _X.,2,Wait(1) exten => _X.,3,Background(msg) exten => _X.,4,Background(3-sec-pause) exten =>

Hi Rosario, I have a problem about CDR: inbound calls are not correctly logged in CDR, it says they are always answered even if they are not. It is very strange since outbound calls and internal calls don't suffer this problem. I'll tell you more: I made Asterisk print the DIALSTATUS variable and it is ok, says BUSY when my internal hardphone SIP is busy. Or maybe it is allright and

Hi, I'm using a HFC card on my asterisk box. I tried to make a conference but it doesn't work. I read on internet to use ztdummy but my server has no uhci (only ohci but it doesn't work) so I cannot use it. I tried zaprtc but after loading the module (it appears when typing lsmod) nothing has changed. Should I buy a x100p to get the right timing? Or there is another solution? TIA

Hi, does anybody know what *zttranscode *module* *is for*?* Thanks!! Giorgio -- _________________________________________________ Giorgio Incantalupo, mailto:gincantalupo@fgasoftware.com FG&A srl - http://www.fgasoftware.com - Voice@Work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

Hi, I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy is working fine but for some reason I cannot. The two machines have the same kernel, motherboard, the same gcc version and the same zaptel 1.4.8. On the second machine zaptel compiles without errors and ztdummy.ko is generated but when I modprobe it I get the following error in messages: rtc: lost some interrupts

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

Hi, I have an Asterisk 1.2.9.1 box connected to an ISDN line via a beronet card (with ports in PTP mode). I noticed a long delay when making outbound calls, more precisely between (taken from Asterisk CLI) "Called 1/XXXXXXXXX/s" and "mISDN/1-u43 is proceeding passing it to SIP/8-5486" I searched on misdn.org but found nothing. I'd like to understand if this delay is

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

I cannot reach voip-info - is it just me or is the site not available ? Julian

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

Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid

I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 32166-1B75-151E-E6DA-E37AEEAA2882@10.100.4.252 for seqno 1 (Non-critical Response) Regards, Arne Morten.

Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

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

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>