similar to: Some errors on sip debug

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

>From my extensions.conf: exten => 888,1,Answer() exten => 888,n,WaitExten(20|m) exten => 888,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) This should: * answer * wait 20 seconds for an extension with music on the background * pass the call to that extension on ${SERADDRESS} What am I doing wrong here? I don't even get the background music while WaitExten is active. I doubt that

Using Asterisk 1.0.9 When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten

Hi all! I need a simple plan for the following: *answer call *wait for 4 digit extension *send call to 4-digit extension entered. I tried the following, but that doesn't work... exten => 998,1,Answer() exten => 998,2,Background(agent-newlocation) exten => 998,n,WaitExten(20) exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) WaitExten obviously does not fill EXTEN with

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

I have some problem to redirect the call from asterisk to ser. 1 thing i am redirecting call to asterisk and then on some extension i want to return the call to ser. Receiving this error: WARNING[23594]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to '"Alex" <sip:xxxxxxx@xxx.xxx.xxx.xxx:5061>;tag=as55a3adbb' --

Dear ALL: My scenario lists below: Assume: UA1 with sip id "1011" And dial number to PSTN is "0939749xxx" There is no modification rule at my CISCO. (It will not change any dialed number) UA1 ==> SER ==> UA2 (SIP to SIP) UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==>

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

Hi. Is there a way to access an object within callbacks defined with set_callback (ActiveSupport 3.0.0-beta4, matching Rails 3)? Use case is as follow: require ''active_support'' class Foo include ActiveSupport::Callbacks define_callbacks :handle_response set_callback :handle_response, :after do |response| # play with the response end

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08

Hi all, I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan but I have the following errors and I'm not able to call anymore. Do you know what can I have to do? My Asterisk is connected to a patton with a SIP trunk. [Mar 22 10:19:03] WARNING[16462]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call

Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '0755ad8f40b9d09d491b635e70bb8905 at

Hello, after upgrade (Debian wheezy) from dovecot 2:2.2.13-1~auto+103 => 2:2.2.13-1~auto+113 I discovered that in my log and all mails stuck in the Queue. I?ve had downgrade v 2:2.2.13-1~auto+103 an everything is fine. What?s wrong? Any ideas or solution greatly appreciated. Thanks. Jul 25 11:03:01 server2 dovecot: lmtp(25638): Fatal: master: service(lmtp): child 25638 killed with signal

Hi, I'm having difficulty getting ices 0.2.2 to see my lame install and my perl install. The lame libraries (v3.91) are installed in: /usr/local/lib/ with the header file at: /usr/local/include/lame/lame.h, but when I run: ./configure --with-lame --with-lame-includes=/usr/local/include/lame/ --with-lame-libraries=/usr/local/ it says it can't find the lame libraries. ldconfig seems

Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line => aaln/2 line => aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485

Hello I am trying to register for fwd from * but having problem and unable to solve it. I keep getting this message *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to '<sip:89699@fwd.pulver.com>;tag=as62a7f29b' NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to

When run asterisk ?vvvgc IT show me this error Asterisk Ready. *CLI> Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'phone@192.168.0.6' timed out, trying again Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from '<sip:phone@192.168.0.6>' failed for '192.168.0.6' Apr 11 08:59:27 NOTICE[81926]:

I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone. When I dial the number for the IP phone off the POTS phone, the IP phone rings. But when I pick up the handset on the IP phone, I get a busy signal and this message on *: Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from svip10@00059002042b-1 Here is the entire session. svip10 is the 1 and

Hi, Folks ... Apologies in advance for the length of this message, but I wanted to be thorough, and provide as much info as I could. I'm trying to figure out a problem in trusted-host authentication using AIX hosts as clients, and a Sun host as the server; either I'm missing something real obvious, or there might be a bug somewhere in some piece of software involved here. -- All of