similar to: SIP response 603 driving me nuts

dear all, what is the meaning of this *Got SIP response 603 "Declined" back from XXX.XXX.XXX.XXX* is it asterisk related issue , because sometimes my outgoing calls working fine , and in a day for 2 to 3 hours it gives me this my provider says its all fine there any one know meaning of this regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current scenario is if I get "BUSY" back from the first provider, I send a busy back to my customer. If I get something like CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt the call. This works

In my dialplan I'd like to send a "603 Declined" message to the user placing the call. I see the commands for the Busy and Congestion, but not the one for the Declined. Any help? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130726/5ac93551/attachment.htm>

I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both

Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;

Asterisk is behind firewall, I'm able to register with the provider. Calls are coming IN OK, but when I try to call out I got: Got SIP response 603 "Declined" back from 81.15.xx.xx -- #Joseph

Hi Guys, I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17. Can anyone shed light ? --- (12 headers 0 lines) --- Transmitting (NAT) to 209.212.93.53:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX Via: SIP/2.0/UDP

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

Hi, I have some problem with my provider. While the sip registration is successful, i intermittently encounter problem in dialing out. I receive 603 Declined error in my Sjphone client. The asterisk log shows line is busy/congestion. Appreciate if help or direction can be provided. Thanks. CK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

Man, I really feel stupid, but after banging my head on a brick wall for several hours ... I need help! I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 5707, and I've got an xlite on 5608. When I make a call from an outside line, I dial SIP/5608. The little blinky light on the GXP that's monitoring 5608 goes, well, "blink blink". :) I then press

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten

Could someone help me with the following code snippet. The results are not what I expect: > Sheet1$Claims[1:10] [1] NA 1 2 NA NA NA NA NA NA NA > Sheet1[1:10,"SubmissionStatus"] [1] Declined Bound Bound Bound Bound Bound Declined Dead Declined [10] Not Taken Levels: Bound Dead Declined Not Taken > Sheet1$Claimsnum <- NA >

Hi all, I dont have any problem, my asterisk is working fine. but on the cli, asterisk keeps saying "Got SIP response 603 "Declined (no dialog)" back from 192.168.0.100". trixbox running on another machine is registered to our server from address 192.168.0.100. whats the reason of this msg? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An

hello everybody and happy new year! I am a newbie in libvirt use, and I use it mostrly throught OpenNebula. Anyway, I noticed that one of the nodes can be a destination of a livemigrate but not the source. This is the environment: 4 blades nicknamed(*) red3, red9, red10, red11, each with 8 Intel E5506 cores (@ 2.13GHz) running CentOS 5 and KVM as hyoervisor. (*) these are names I

I'm trying to connect to vbox hypervisor on an Ubuntu 10.04 machine through libvirt java binding (libvirt-java-0.4.6) by simply invoking: Connect conn = new Connect("vbox:///session", false); but I got this exception: libvir: warning : Failed to find the interface: Is the daemon running ? libvir: Remote error : unable to connect to '/var/run/libvirt/libvirt-sock':

I just noticed this upon startup since updating from 15.6.1 to 16.0.0 - do any of these matter? [Oct 18 12:12:18] WARNING[4489]: loader.c:2228 load_modules: Some non-required modules failed to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: res_pjsip_transport_websocket declined to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: cdr_sqlite3_custom declined to