similar to: Regarding error in asterisk 1.6.2.16....

Hi every one, I am using asterisk version 1.6.2...... i did not install mysql data base and when i tried to register a client from SIPp xml file..... it is registered.... My questions are 1. where can i find that registered client? 2. when i type the command "core stop now" it exists and the registered users are not shown why this is happening? 3. Is it compulsary

Hi all, i am running asterisk by using command asterisk -r, asterisk -vc ............ when i want to come out of asterisk it not getting exit or quit from the shell.... i have tried soo many options like... stop now stop gracefully exit stop quit its not working stillll can any one tell me what would be the problem with this? please help me ... :( with regards, viswavardhan

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 just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no problems and while everything seems fine i get these message at startup and than all is well. Should I be worried or do i need to let the team know about this? Also, is not finding "/dev/dahdi/transcode" a problem I should be worried about? And lastly conf2ael always segfaults when I try to run it. it did run once

Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i

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 all, My experiment scenario is like this: SIPp Uac -----------------------------> ASTERISK SERVER---------------------------------->SIPp uas 1. when i had registered bob with this command ./sipp -sf register_client.xml -inf register1.csv -i 192.168.1.6:5060 192.168.1.6 -p 5061 -m 10000 it has registered.... If i want to register another client alice with same command

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.

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:

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.

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

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

I'm getting a lot of these on the CLI : [Jun 10 13:41:36] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:37] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:38] WARNING[4286]:

When I try to make a call using the Mediatrix 1204 is showed on the CLI: -- Executing SetCIDNum("SIP/2009-4df1", "1111") in new stack -- Executing Dial("SIP/2009-4df1", "SIP/2217008@192.168.199.5") in new stack Aug 11 15:14:10 WARNING[1211108144]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x81 40c5c (len 794) to 192.168.199.5 returned -1: Operation not

On 08/29/2018 09:42 AM, Carlos Rojas wrote: > Hi > > Probably somebody is trying to hack your system, you should block that > ip on your firewall. > > Regards > > On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I'm getting invites to very high ports every 30 seconds from a

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

Greetings, Since the past week I've started receiving the following warnings on my asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself with x-lite/x-pro/eyebeam clients as well as sipura devices. All of them have qualify=yes in their settings. Jul 18 22:52:01 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address

I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up:

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040108/748d21b3/attachment.htm -------------- next part -------------- Hello I downloaded latest CVS version and got following error messages while registeration. ( CVS Version Asterisk CVS-01/08/04-14:20:34) As a result IP Phone don't register with the Asterisk. Is it broken ? How can I

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",