similar to: CPU load 100% when SIP register

----- "Chandrakant Solanki" <solanki.chandrakant at gmail.com> escreveu: > Hello > > I have successfully compiled OSLEC for echo cancellation for DAHDI > channel. > > Is there any way to do echo cancellation for SIP Channel. > > Is any, please suggest me.?? > > Thanks in advance.. > > -- > Regards, > > Chandrakant Solanki Short

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'

I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout):

Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config sip.conf I don't think it's permissions: ls -ld /etc/asterisk /etc/asterisk/sip*

I can not register via sip to iptel or sipgate. i do not unterstand why.. but i am new to asterisk. Iam behind a susefirewall2 but asterisk even do not register if it shut down. No answer seems coming back. thx for help. nico here is my config if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr =

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc77 at 178.19.90.240>' failed for '78.119.140.190:5076' - Wrong password [Aug 15 22:04:13] NOTICE[30098]:

Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr) IRIS

Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

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

Ok, here comes part two of the log quiz, this time SIP not MGCP: WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER that isn't a register This is most probably cause by registration of * with FWD. Cheers, Philipp

Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate

ok, solved the firewall issue. A first test call worked fine. Another one now still gets disconnected after 32s. But in FW there are no blocked packets anymore?! And I don't understand why the registration to the same IP and same Port is working, but not later transmission of further SIP packets? that doesn't sound logical to me. What do you think? regards, andre

Hi all, I have a strange issue, with a some kind complicate architecture... A router of our internet provider is in front of another bintec rs353j router, at which my freepbx installation is located. However, NAT etc. seems to work fine. BUT: Something is not working...: When registering my sip-trunk towards my provider (3 different providers, all behave comparable), everything works at first.

Hi, Since I upgraded from Asterisk 1.2.18 to 1.4.17 I've been experiencing high CPU utilization from the chan_sip module. I've notice the more sip peers I have loaded, the higher the CPU load goes when there are no active calls. I am currently using a Pentium 4 3.0Ghz with CentOS 4 Kernel 2.6.9-42.0.2.EL. I currently have 1558 sip peers loaded in Asterisk and the current CPU load is

Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr)

Hello, I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands) >From the moment i switched all inbound calls are terminated after aproximatly 1 minute. The provider tells me it's not their issue since I have no other configuration than all their other users. What can I do. I removed all asterisk functionality by forwarding the inboud call directly to a local

Hi all, I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall related, but I'm unsure. A registration to Sipgate is established successfully: <Registration/ServerURI..............................> <Auth..........> <Status.......> ==========================================================================================

Is there any reason why we wouldn't want to add a setting for each REGISTER => line in sip.conf that allows us to set refresh time on a per-proxy basis? If not I thought I would try to add that functionality to chan_sip. Please tell me if there's some reason why we wouldn't want to do this, (e.g. it's coming in a future CVS or in chan_sip2 or it will horribly break the code