similar to: Oops, a seg fault =(

Just working on Configing up Voicemail and now that I have got it working and configed and answering the way it should be I have another challange. on the * CLI> I get this -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg0000 format: wav49, 0x8133390 -- x=1, open writing:

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

Asterisk Project Security Advisory - ASA-2007-014 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Stack buffer overflow in IAX2 channel driver |

Asterisk Project Security Advisory - ASA-2007-014 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Stack buffer overflow in IAX2 channel driver |

Hi, I'm running Asterisk 1.0.7 and would like to add Speex support. I downloaded Speex 1.0.5, installed and recompile Asterisk again. Now trying from X-Lite to connect using Speex but getting lot of weird erros on Asterisk console: Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein: Out of buffer space I was trying to setup Speex on my second Asterisk server and wanted to

Hi, I'm sorry to bring this up again, but I have been googling forever and whatever solutions are offered don't work for me. I am using x-lite (the latest build) and trying to use Speex. When I do call from the x-lite to another SIP phone or PSTN (through Cisco gateway) My asterisk fills up with this message: WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space The

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

I'm running the latest version of cvs (not stable), I'm not sure what the other end is running and if this has been fixed or not yet, however I was playing round with onhold earlier, the call went to onhold, and came back from it, then 2 seconds later was hung up unexpectedly, below is what was on console... -- Started music on hold, class 'default', on

Hi, I upgraded my local Asterisk (the last version was quite old), and since then, whenever anyone tries to call me via SIP/IAX thru my external Asterisk, they get "403 Forbidden" as soon as I pick up. I have no trouble picking up when someone calls via PSTN. Basically, my phone (Firefly softphone) will ring when they call, but will disconnect as soon as I pick up. It won't even

Hi, i am using iax client and when i try one of my extension that play MusicOnHold() it give me this error, who have an idea about this - Executing MusicOnHold("IAX2[1@1]/1", "") in new stack Jun 4 15:36:37 WARNING[1217602880]: chan_iax2.c:2838 iax2_send: timestamp is 0? Jun 4 15:36:37 WARNING[1217602880]: channel.c:1445 ast_prod: Prodding channel 'IAX2[1@1]/1'

Hi, I was messing around with FireFly last night and got asterisk to crash hard. It looks like the bug is a division by zero in chan_iax2.c. I reproduced it and here are some infos I got from gdb: [Switching to Thread 245775 (LWP 23251)] 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 2896 int diff = ms % (f->samples / 8);

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

Hi, For 3 days uptime serve about 40 client Windows workstation with traffic average 50 Mbps - 80 Mbps (Video streaming) running on FreeBSD system with 16 GB RAM already installed. # uname -smrv FreeBSD 12.1-STABLE FreeBSD 12.1-STABLE r364492 GENERIC amd64 # pkg info samba\* samba410-4.10.17 I got this produce error in /var/log/message, any clue for this problem? Sep 14 10:20:15 BEC-STG-P1

Hi, I am having problems using pam_winbind to log in as a user in a trusted domain. The arrangement is that Samba is joined to a local domain DOMLOCAL which has a trust setup with DOMREMOTE. getent passwd/group correctly enumerates users and groups from DOMLOCAL. If I try getent passwd for the DOMREMOTE account no result is returned. pam_winbind has a requirement that the user is a member of

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

Dear Samba-users, I cannot manage to make samba read my logon.script for windows clients where basicaly networkdrive-mapping is done. Samba complaines about not having permission... Error: switch message SMBntcreateX (pid 28286) [2004/12/03 11:27:31, 3] smbd/dosmode.c:unix_mode(110) unix_mode(logon.bat) returning 0744 [2004/12/03 11:27:31, 3] smbd/open.c:open_file(173) Error opening file

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

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",