similar to: Failure to write to tcp/tls socket

I?m trying to use Playtones to have a tone played periodically throughout phone calls. Unfortunately, I can?t seem to get PlayTones to work. I never hear the audio tones. Here is the output on the Asterisk console. -- Executing [19525553312 at proxy-dial:2] PlayTones("SIP/testphone-00000032", "1400/500,2000/5000") in new stack [2014-10-30 14:28:31] WARNING[23154]:

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Can anyone tell me why I keep getting these messages from IAXTEL? It does appear to register since I get lines like this: 2005-04-30 04:26:42 VERBOSE[1644]: -- Registered to '69.73.19.178', who sees us as 67.182.152.242:4569 But what is this? I don't think IAXTEL is working for me, since I can't dial 800 #s through it when I copy the iaxtel.com instructions. 2005-05-20

Hi I have created meetme with 3 user. When i going to mute user it gives following error.. *Asterisk Version : 1.6.2.6* -- <SIP/52987-00000040> Playing 'conf-muted.gsm' (language 'en') [Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid GSM data (1) [Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 [Jul

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0 [Jun 2 17:08:28] WARNING[21652]:

Hi, I just installed asterisk 1.4.0 on my mac. I compiled from source with no issues. I installed the sample config files, and basically just added a register line to sip.conf (to register with a Free World Dialup account). Then I called my asterisk system from a different computer (using x-lite softphone on windows xp, registered to an ekiga.net account). Asterisk answers, and I can hear the

On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find

Hi, I got ast. 1.6.0.10 working for a few weeks without a problem. A few mins ago..I got the following msgs on ast-cli and asterisk service crashed. I coudlnt find anything that might cause this problem. Any ideas?? [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0

http://bugs.freedesktop.org/show_bug.cgi?id=20511 Summary: No sound with remote pulseaudio Product: swfdec Version: unspecified Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: plugin AssignedTo: swfdec at lists.freedesktop.org ReportedBy: observer1

Hello all! Actually I use dual boot PC with CentOS4 (kernel 2.6.9-11) and WinXP (HDD 20GB, 512MB, PentiumIII 800MHz), but I have the following problem doing the image with g4u ----BEGIN ERROR MESSAGGE---- wd0: (uncorrectable data error) 8951MB 1.59MB/s wd0d: error reading fsbn 18333440 of 18333440 - 18333567 (wd0 bn 18333440; cn 18187 tn 14 sn 62) retrying wd0: (uncorrectable data error)

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

CentOS Errata and Bugfix Advisory 2018:2754 Upstream details at : https://access.redhat.com/errata/RHBA-2018:2754 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 8bbf735162d71c0ed1ecc5527f08bfc9bb9e61534a6e97df61b17824530e9edb libgfortran4-8.2.1-1.3.1.el7_5.i686.rpm

Hi all, Our kernel is 2.6.32-358.14.1.x86_64, recently dozens of them panicked, since it's been OK for a long time and the problem emerged all of a sudden, I'm not sure if an upgrade caused this problem. Here's what I got from backtracing: PID: 8136 TASK: ffff8803341aead0 CPU: 2 COMMAND: "" #0 [ffff880028283610] panic at ffffffff815286b8 #1 [ffff880028283690]

Dovecot version: 2.3.1 (happens with 2.3.x too) OS: CentOS 7 64-bit Mar 28 16:29:24 lmtp(30383): Panic: file lib-event.c: line 182 (event_pop_global): assertion failed: (event != NULL) Mar 28 16:29:24 lmtp(30383): Error: Raw backtrace: /usr/lib/dovecot/libdovecot.so.0(+0xcc7a4) [0x7fac7f5177a4] -> /usr/lib/dovecot/libdovecot.so.0(default_fatal_handler+0x2a) [0x7fac7f5177ea] ->

virt-v2v is not an option for me as all I have are the VMDK images and don't have the VMware software to covert to OVA first. I really don't understand what is going wrong. I also found https://access.redhat.com/site/solutions/20511 and that still doesn't work. Both /etc/fstab and grub.conf refer to the LABEL and not any device name. In the boot process I see it explicitly says it

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also