similar to: 1.6.2-RC1 question

I get the following warning when starting Asterisk 1.4. Does anyone know what these mean, and/or how I can get rid of them? [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend. I'm getting: [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]:

I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

I've got a question about codec_dahdi witrh a system running Asterisk 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to route calls between different PRI connections, so no transcoding between codecs is happening as far as I am aware. 1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? 2) I'm

I have a CentOS release 4.7 box running asterisk-1.4.26.1 with dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0 I regularly get these messages, is this something i should be worried about? [Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Aug 21 01:05:07] ERROR[4343] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory [Aug

I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function

I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 installed from the asterisk.org and digium.com repositories. I have Asterisk starting (service asterisk start) but see errors about dahdi in /var/log/asterisk/messages. ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory Linux-Vservers don't allow, under normal circumstances,

I haven't tracked this down to anything on my system yet, but has anybody else upgraded to 1.4.0b4 (from 1.4.0b2) and found that asterisk core-dumps on startup? The last few lines in messages before dump are: [Dec 16 10:44:03] WARNING[7958] translate.c: plc_samples 160 format 6 [Dec 16 10:44:03] NOTICE[7958] chan_agent.c: No agent configuration found -- agent support disabled [Dec 16

Hi A question, i have upgraded a beta serveur from Asterisk 1.6.1 to 1.6.2 and now all SIP Relatime user are rejeted : [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21639 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21658 handle_incoming: Ignoring SIP message because of retransmit (REGISTER Seqno 44199, ours 44199) [Oct 5

Hi. So, I applied the patch, works, but I could not figure out a fail2ban regex which will hit that line, have you got one I can use? Thanks. On Thu, 30 Aug 2018 11:03:08 -0400, sean darcy wrote: > > On 08/29/2018 09:33 PM, John Covici wrote: > > OK, Thanks. I have a couple of questions -- the line numbers do not > > match exactly, so can you tell me a couple of lines before

Hi guys, I just ran into a funny issue here. I'm trying to virtualize our asterisk pbx onto vmware esxi. Here's a quick glance of the system: * Ubuntu 9.10 i386 with linux-rt kernel (to get 1000Hz timer) everything up2date. * Asterisk 1.6.2.6 If I run asterisk using the debian init script in contrib/init.d/, top shows asterisk is using 99.x% CPU doing nothing. If I run asterisk with

As of today, during startup I get lots of the following: ERROR[2704] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data Does anyone know what it means? This is with Asterisk 1.6.0.9.

Hi every one, When i run asterisk i am getting error as ERROR[2109]: chan_sip.c:3963 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data ............ i am unable to understand what it is and how come.......... can any one help me regarding this issue.......... Hope i get a positive reply...... With Regards, viswavardhan --------------

Hi when i use MeetMe, i have this errors: app_meetme.c: Unable to open pseudo device Where is the problems ? i have too warning and error into my logs: [Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. [Nov 1 07:26:17] WARNING[18544] config.c: Realtime mapping for 'iaxpeers' found to engine 'mysql', but

In a (futile?) attempt to get rid of warnings, I have this: [Nov 30 10:39:49] NOTICE[68467]: loader.c:937 load_modules: 149 modules will be loaded. [Nov 30 10:39:49] WARNING[68467]: utils.c:1427 __ast_string_field_init: trying to reset empty pool (5 times more) SIP channel loading... (5 lines of AEL loading) [Nov 30 10:39:49] NOTICE[68467]: pbx_ael.c:149 pbx_load_module: AEL load process:

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

OK, Thanks. I have a couple of questions -- the line numbers do not match exactly, so can you tell me a couple of lines before and after the line in question? Also, when will this be logged, if its only during sip debug, I need to change it to log when I can see it more readily. Thanks. On Wed, 29 Aug 2018 20:31:15 -0400, sean darcy wrote: > > On 08/29/2018 08:07 PM, John Covici wrote:

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

Hi, After a few attempts, I've managed to grab the files from CVS and build it on a rh redora box I've setup especially for Asterisk. Firstly, we're new to the asterisk scene, so please excuse any "lame" questions which may follow.. We're a new voiptalk.org customer. We have purchased the voip phones (budgetone 102's) and set aside a little box to run Asterisk on.

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and