search for: 603

...pported. However, I can't find the download for this version. Is the support official and where can I download it. Thanks / regards, Ivan Wong Database Administrator e2Open Inc. (www.e2open.com) Suite 34.03, Level 34, Menara Citibank, 156, Jalan Ampang, 50450 Kuala Lumpur, Malaysia DID: +603 2776 6397 Tel: +603 2776 6300 Fax: +603 2712 9112

...}|1) [a2billing] exten => _X.,1,DeadAGI(a2billing.php|1) exten => _X.,2,Hangup(34) ;============================= A2Billing authenticates and routes the call properly, but when the termination gateway for the destination dialed by the customer rejects the call, my Asterisk box sends "603 Declined" to the customer. It also happens when A2Billing doesn't find any route for that destination, in which it should return "404 Not Found", but returns "603 Declined" instead. I tried to force every rejected attempt with "503 Service Unavailable" putt...

I've been running Samba 2.2.2 for a while as a domain controller on my mostly Windows network. It's been working great up to this point. Anyway, I just put together a new Linux server and installed 2.2.2 on it as well. I copied the conf file and pretty much mirrored everything on the existing server, changed the paths and the "workgroup" field in the conf file to match the new

...}|1) [a2billing] exten => _X.,1,DeadAGI(a2billing.php|1) exten => _X.,2,Hangup(34) ;============================= A2Billing authenticates and routes the call properly, but when the termination gateway for the destination dialed by the customer rejects the call, my Asterisk box sends "603 Declined" to the customer. It also happens when A2Billing doesn't find any route for that destination, in which it should return "404 Not Found", but returns "603 Declined" instead. I tried to force every rejected attempt with "503 Service Unavailable" putt...

...ent 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 great as expected. However, One of my SIP carriers likes to send back 603 "DECLINED" inplace of 503's. Asterisk ${DIALSTATUS} treats this as "Busy". Can I change how asterisk interprets a 603 Declined? So it treats is as "CHANUNAVAIL"? The obvious quick fix is to change my "Busy" option to attempt another carrier before...

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 HTM...

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; if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause))) transmit_response_reliable...

...quite some time to find a solution for the following problem. I have a data frame which is organized by block and trial. Each trial is represented across several rows in this data frame. I'd like to extract the first x rows per trial and block. For example block trial x y 1 1 1 605 150 2 1 1 603 148 3 1 1 604 140 4 1 1 600 140 5 1 1 590 135 6 1 1 580 135 7 1 2 607 148 8 1 2 605 152 10 1 2 600 158 ..... Selecting only the the first two rows per trial should result in block trial x y 1 1 605 150 1 1 603 148 1 2 607 148 1 2 605 152 The data I am dealing with a x-y coordinates (samples) f...

...when it fails. Thank You, Steven BerkHolz Board member of www.glimasoutheast.org ref: from SIP Phone (I think these are the second invite after it is hung up) 2006-OCT-20 17:49:52 GMT +++ Current Timestamp +++ 2006-OCT-20 17:19:47 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:56:37 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:50:00 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:45:38 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to RE...

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...

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20070221/e599edad/attachment.htm

Hello list. Is it possible in the Asterisk dialplan to send a 503 Service Unavailable of 603 Decline after having answered the call with Answer() in the dialplan ?? Suppose that I first want to check the call in a MySQL-database, while I put some MoH, and then let the call go through or "send" some error to my ITSP where the call comes from. I know there is something like 'e...

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/...

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 XXX.XXX.XX.XXX:5060;rport=5060;received=74.96.44.239;branch=z9hG4bK-24c7d466 From: <sip:XXX.XXX.XX.XX>;tag=a21dc3d8dd92817bo0 To: <sip:XXX.XXX.XX.XX>;tag=as7b187bff...

Hi All, Have any way to asterisk forward the 487 Request Cancelled in SIP TO SIP call? In a SIP to SIP call when the called peer B send 487 to Asterik, Asterisk return to calling peer A 603 PEER A ASTERISK PEER B | INVITE ------------>| | |<------------TRYING| | | | INVITE------->| | |<-------TRYING| |...

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20101014/8...

...001>uname -a Linux x335-142 2.4.9-e.25smp #1 SMP Fri Jun 6 18:11:40 EDT 2003 i686 unknown Appreciate any feedback. Thanks / regards, Ivan Wong Database Administrator e2Open Inc. (www.e2open.com) Suite 34.03, Level 34, Menara Citibank 156, Jalan Ampang, 50450 Kuala Lumpur, Malaysia DID: +603 2776 6397 Tel: +603 2776 6300 Fax: +603 2712 9112

...rmally, 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 -- SIP/103-00000e89 is ringing -- Got SIP response 603 "Decline" back from 192.168.215.104 // (104 is the IP of SIP/103) == Spawn extension (cola-radio2, s, 4) exited non-zero on 'DAHDI/6-1' -- SIP/103-00000e89 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing [103 at AgentDialFromQ:2] Hangu...

...ll to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to transcode g729. What is asterisk doing? Asterisk decides to stop the call at all: - Sends cancel to provider and 603 decline to internal caller. What would have been correct? It would have been correctly to switch to alaw as provider offered it, too. Workaround: My workaround is to disable g729 and things are working fine again for me (for this special case). I'm seeing here two basic problems: 1. Aster...

http://bugzilla.mindrot.org/show_bug.cgi?id=603 Summary: configure: error: OpenSSL version header not found Product: Portable OpenSSH Version: 3.6.1p2 Platform: UltraSparc OS/Version: SunOS Status: NEW Severity: major Priority: P2 Component: Build system...