similar to: sip debug command?

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

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

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

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

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",

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54

Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922

Hi folks, thanks for your help with my last question re: japanese FXO. It doesnt sound very compatible so I will use a SIP FXO gateway then. Untill I find one, im just trying to get my 2 cisco SIP phones talking to my * server. just as a learning experience for now. heres what I have so far: 2 Cisco 7960's both using DHCP and both registering with my SIP proxy server (Brekeke OnDo on

Hi, I've recently got * working (thanks Clive and list!) at home. We have 2 PSTN lines connected via X100P cards. I've got 3 x SIP phones (2 are Budgetone, the other is a Tecom SIP). One of the lines is our standard home line, the other a business line. Presently I've got * set so you dial 91<tel no> and 92 <tel no> to select which line to dial out on. I should probably

I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102

Just upgraded from 0.7.1 to the latest CVS version yesterday. This introduced a slew of warnings on startup. About 20 or 25 of the first, then 5 of the second: Jan 23 10:58:49 WARNING[8201]: chan_sip.c:446 __sip_xmit: sip_xmit of 0x80ef064 (len 461) to 0.0.0.0 returned -1: Invalid argument Jan 23 10:58:55 WARNING[8201]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call

I'm getting the following errors out of my asterisk console: Jul 14 15:12:12 WARNING[40966]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 5d33ce4422e629921875e8a7609c1603@0.0.0.0 for seqno 141 (Critical Request) Jul 14 15:12:26 NOTICE[40966]: chan_sip.c:3902 sip_reg_timeout: Registration for '451411@fwd.pulver.com' timed out, trying again Jul 14 15:12:26

Hello *'s, Every time I started my asterisk error shows: chan_sip.c:Got 200 ok onRegister that isn't a register. Failed to authenticate in Register to <sip:adnan.007@iptel.org>;tag=asfeaa71f Maximum retries exceeded on call xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx@192.168.10.189 for seqno 107 (Request) Registration for adnan.007@195.37.77.99 timed out ,trying again.

Hello, I have a strange situation with my asterisk 1.8.7.0 version. I compiled as usual everything seems to be ok but from time to time when i look on my console i get the following error message: [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write

Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI> NOTICE[1158921008]: File chan_sip.c, Line 5394

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

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

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]: