similar to: 403 Forbidden since upgrading

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 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

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

Hello, I have upgraded to 1.0.4 version of asterisk. After that asterisk crash every time On receiving an call from iax2 trunk to musiconhold application. SIP calls to MusicOnHold is however working. I already upgraded to 1.0.5, but the problem still Remainig. Any idea ? Iax2 : call proceding : Jan 25 17:29:40 DEBUG[9997]: pbx.c:1261 pbx_extension_helper: Launching 'WaitMusicOnHold'

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 |

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

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

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 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);

I''m getting a 403 Forbidden error when I try to access the rails welcome page. I''ve followed the instructions for implementing rails on Apache2 at http://wiki.rubyonrails.org/rails/pages/Tutorial using an alias, but isn''t working. Can anyone help me out? Thanks. -Ofir -- Posted via http://www.ruby-forum.com/.

Hi list I'm trying to get access to the Compiz forums, but I am greeted with a "403 - Forbidden"-page. Is this a known issue? Regards, Rune

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

Hi all, I have configured Asterisk using database. The real time is working pretty well. That is asterisk is picking up details of peers and the extensions as well properly from the MySQL database. -- Executing Wait("SIP/bharat-f720", "2") in new stack -- Executing NoOp("SIP/bharat-f720", ""Welcome to Asterisk"") in new stack

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

Hi all, I have been worriyng and googling a lot but I can't find my mistake. I am trying to regiter an X-Lite Softphone to Asterisk, but I am getting a SIP/2.0 403 Forbidden response: SEND TIME: 10157385 SEND >> 10.100.249.12:5060 REGISTER sip:10.100.249.12 SIP/2.0 Via: SIP/2.0/UDP 10.100.249.86:5060;rport;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22 From: Tester

trying to set up and configure a polycom soundpoint ip 500 phone, when trying to get it to register with sip, i get the following message Sip read: REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58 From: "138polycom" <sip:5622832456@67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC To:

Download V 0.4 here http://sourceforge.net/project/showfiles.php?group_id=123387 burn it to an .iso install into asterisk box (be warned it deletes everything on the hard drive but this is what you want right :) it will automatically install Asterisk AMP FOP and Web Meetme read the FAQ here http://asteriskathome.sourceforge.net/faq.html basically if you are using a X100P all you need to do

Hi guys, I'm using Centos 3.5 with Apache-2.0.46. i linke my mrtg from /var/www/mrtg to /var/www/html/mrtg so i did the command ln -s /var/www/mrtg. it worked fine last week but when i checked the mrtg today it say 403 forbidden. Forbidden You don't have permission to access /mrtg/ on this server. ------------------------------ Apache/2.0.46 (CentOS) Server but when i tried to link

Hi, I have two softphones connected to an Asterisk "stable". I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a "403