similar to: Codec Mismatch

[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten => 6001,1,Answer exten => 6001,2,Background(blahblah) exten => 6001,3,Ringing exten =>

Hi list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

Hi Richard, Thank you for the reply Correct, I did mean 13.15. I set no optimize and better backtrace through "make menuselect" and the output is now [Aug 28 21:41:16] ERROR[17171][C-0000392d]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x21962b0 (0) Got 26 backtrace records #0: [0x61923f] main/utils.c:2475

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I've recently setup a small load test against an instance of Asterisks. I've tested on asterisk 13.5 and 14.6 with the same results. I am using PJSIP. My dial plan is, [test] exten => 1001,1,Answer exten => 1001,n,MusicOnHold(15) exten => 1001,n,Hangup I am using SIPP to test. I can share XML if desired but it simply waits on the line while music plays for 8

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, This patch fixes a couple of segfaults in music-on-hold, frame smoother routines and channel allocation in Asterisk. Mark, feel free to apply it in CVS (if approved). Regards, Michael. -------------- next part -------------- Index: channel.c =================================================================== RCS file: /usr/cvsroot/asterisk/channel.c,v retrieving revision 1.25 diff -u

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

Hello, I'm Using Asterisk 1.8.11.0 on Debian Squeeze. I was experiencing problems with ${SMSSRC} being blank, so I applied this patch: https://issues.asterisk.org/jira/secure/attachment/42026/sms-sender-fix.diff but otherwise everything is standard. As the subject says, if I am making a call through the phone when an SMS is received, the bluetooth connection drops and the call ends. The

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Stuck with TLS transport, I have 2 phones (both in local network for tests) one connected with up second with tls when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 <?>: tlsc0x7f143012 TLS connect() error: Connection refused [code=120111] pjsip log: -- Called PJSIP/601/sip:601 at 192.168.1.55:5075;transport=tls <---

Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past

I''m pleased to announce the release of Cross Site Sniper 0.2. Cross Site Sniper is one more addition to the ever growing list of tools that attempt to provide a convenient and DRY method to protect Rails sites from Cross Site Scripting (XSS) attacks. There are many plugins and tools out there that attempt to address this issue, but none of them met my requirements. So, I created

On Sat, Jun 10, 2062 at 01:16:03PM -0600, jd1008 wrote: > On 06/12/2015 01:01 PM, Gordon Messmer wrote: > >As far as cookies go, you're even further from the truth. A script can > >only access cookies whose domain matches the origin of the script. > > Your final line is not true. Its technically true, however, XSS attacks can get around that restriction, which is why

On 06/13/2015 01:05 PM, jd1008 wrote: <<<>>> > Mark, please be aware that noscript has also a whitelist > that is not viewable by the user. > The whitelist tab does NOT list the hidden white listed > entries. and you know this how? i do not really believe there is a 'hidden whitelist'. it is more like there are sites that are used to check on sites you

Welcome to the first Puppet Dashboard maintenance release of the new year. This release includes a security update to address CVE-2012-0891, a XSS vulnerability discovered by David Dasz <david@dasz.at>. We have classified the risk from this exposure as moderate. All Puppet Dashboard users are encouraged to upgrade when possible. Puppet Enterprise users should visit