similar to: Got SIP response 603 "Declined" back from 81.15.xx.xx

I have one Asterisk box registering to another via SIP and on the registar console I keep getting: -- Got SIP response 603 "Declined (no dialog)" back from xxx.xxx.xxx.xx Anyone know how to turn off this "feature"? -------------- 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======

Hi to all, I have a strange behavior in my asterisk server. I have a queue for 5 agents, the calls enter the queue an go to the agents normally, but if I need to transfer or dial directly to an agent extension that is already in a call, the pbx hung up the actual call (not the transferred call). This is what I see in the log. Called 103 -- Agent/103 is ringing --

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

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;

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======

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:

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>

Hi, I have an Asterisk 1.8.2.3 installed (public IP) with a peer (Polycom IP601) installed behind NAT. When the peer makes a call, it's working without any problem. But when a call is coming back, it ends up with a Got SIP response 400 "Bad Request" back from xx.xx.xx.xx where the xx.xx.xx.xx is the public IP of the peer. And the call drops to the voicemail (congestion at peer

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:

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 everyone, Since three weeks ago, we have been getting A LOT of 603 Declined calls from iCall. I called a few times and their support is either non-responsive (they never call back) or can't fix the issue. I am wondering if everyone else is experiencing the same thing or is it because we recently upgraded from Asterisk 1.6x to Asterisk 1.8x and there is something that is causing this. This

I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes

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

Hello, I have a situation where a call comes in to my Asterisk server B. This call comes from another Asterisk server A. I want to tell to this server A why my server B hangs up. So just before hanging up, I add a custom SIP-header : exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten => s,n,Hangup() But I notice that this extra SIP-header is not send within the SIP-reponse :

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

Hi, Im trying to register to a SIP provider that told me that they only need to authenticate using IP. the following string generates response 405 register => asteriskIPaddress@SIPproviderIP:5060 doing the following is not alowed by asterisk register => @SIPproviderIP:5060 any ideas? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/

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 &

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

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