similar to: 603 Declined > Dialstatus Busy

Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? I've got a new box running 13.3.0 with the exact same issue. For those that don't read the link. I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These are loaded into asterisk without

Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

Hey everyone, Currently, I've got a PBX that is emailing me on call failures to an international SIP provider of ours. I'm doing this with exten => 1,1,System(mail -s "Call from ${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}" nick at flhsi.com < /dev/null) This works fine, However it's a little lacking. For Instance, Our INTL SIP

Hello All, Looking for a little guidance on Real Time Pattern Matching. We are attempting to block outbound 411 via when someone dials NXX-555-XXXX, The must common being NXX-555-1212. However, We have some outbound providers that consider any call to NXX-555-XXXX a directory assistance call. So simply making my pattern _NXX5551212 doesn't work. So as you can see from the lines

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

Hello, is it possible to disable the CDR record creation for all dialed parties? From my limited testing it looks like CDR_PROP(disable) is effective only for the first party (the one specified before the first ampersand in the Dial application argument) and I can't find any way to disable it for the other ones (I think the CDR in question is written after the Dial completes). Is it by design?

I currently have in extensions.conf: exten => 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,n,Monitor(wav,${CALLFILENAME},m) exten => 106,hint,SIP/106 exten => 106,Macro(stdexten,106,${HINT}) When I called x106 this was logged: -- Executing [106 at voicemenu-custom-4:1] Set("DAHDI/7-1", "CALLFILENAME=_xxxxxxx") in new stack --

I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -------------- next

Hello, I read on the wiki : Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using the destination channel, not the source channel. But when I use this in my dialplan, this 'variable' is empty. Dialplan : exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})}) exten =>

Hello R user, I have several data frames with >100 columns and I did a linear regression over time of each column df1.lm <- lapply(df1, function(x) lm(x~year)$coeff[2]) that worked fine and I get slope of each column oder time - until I divided df1 by df2 df3 <- df1/df2 > df3.lm <- lapply(df3, function(x) lm(x~year)$coeff[2]) Error in lm.fit(x, y, offset = offset, ...) :

Greetings, Recently a performance regression in chan_sip was discovered in Asterisk 1.8. The regression is caused by chan_sip setting MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each response received on a channel. That feature has been made optional in the latest 1.8 SVN code, but is currently still enabled by default. After some internal discussion, we decided to consider disabling

Dear friends, is there anybody who can help us, I've the same problem with the Kickoff Time. I also found no possibility to change it... Heinz -------- Original-Nachricht -------- Betreff: [Samba] Kickoff time expired, how to change? Datum: Mon, 12 Nov 2007 10:35:32 -0700 Von: Tim Donnelly <tim@coalliance.org> An: <samba@lists.samba.org> I am attempting to move my

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise -

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>

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'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 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