search for: causecod

...=> s,1,SetVar(CALLFILENAME=/var/www/recordings/${TIMESTAMP:0:8:7}/${UNIQUEID}) exten => s,2,AGI(recordstart.py,${ARG1},${CALLERIDNUM},${CALLFILENAME},Ind) exten => s,3,DIAL(ZAP/g2/${ARG1},70) exten => s,4,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME}, ${CAUSECODE}) exten => s,5,hangup exten => s,104,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME}, ${CAUSECODE}) exten => h,1,stopmonitor exten => h,2,SetVar(CALLFILEDIR=/var/www/recordings/${TIMESTAMP:0:8:7}) exten => h,3,System(/etc/asterisk/agi-bin/filexfer ${CAL...

Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.

...;Hi, all<br /> <br /> Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked?<br /> <br /> Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.</blockquote><div>š</div><div>I only know of the SIPAddHeader application which lets you add headers when used before Dial, so I don't think you can do this currently.š</div></div><div>...

...com.net --[IAX2]--> gate.tubby.org If I ran debug on the central box (asterisk.thorcom.net) I could clearly see the call coming in and being placed on gate.tubby.org but it was being rejected with the message: [Oct 29 19:47:16] WARNING[16974]: chan_iax2.c:770 iax_error_output: Expecting causecode to be single byte but was 2 [Oct 29 19:47:16] WARNING[16974]: chan_iax2.c:7450 socket_process: Call rejected by 193.82.116.194: No supported codec found Now, over at gate.tubby.org a 'tcpdump' clearly showed the exchange of IAX packets, but enabling debug on IAX showed nothing!? I...

...wrote: > On Tue, 2004-06-22 at 18:43, Simon Brown wrote: >> This should be listed as a bug - it is not logical to go to busy, >> when in fact the extension is unavailable. > > I think the whole idea of "busy" or "unavailable" is flawed. > Asterisk sets ${CAUSECODE} with the cause of the call being > cleared. You can use this to determine what you want to do. > For exmaple if the cause code indicates "unallocated" then > you should give the caller some indication that they number > they dialed is disconnected or no longer in service. &...

Okay, I think I need a sanity check here - If I call a person that's on the phone, I should get a busy signal. Now more specifically, a call comes into the pbx via PRI. The destination dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user busy) so that the originating end can play a busy tone, correct? -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140709/8b98a0f1/attachment.html>

Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong number" to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides "wrong number", I would especially like to send 302 temp moved with a specified address t...

...pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only "normal clearing". Interestingly on Box A the NoOp Output of PRI_CAUSE and CAUSECODE after the dial attempt to an unallocated number is also empty. How to access/query the reason for a failed ISDN dial attempt from the dial plan, I expected at last PRI_CAUSE to be filled. Is it possible at all with asterisk - what modification to my attached configs would you suggest? Which vers...

...;Hi, all<br /> <br /> Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked?<br /> <br /> Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.</blockquote><div>š</div><div>I only know of the SIPAddHeader application which lets you add headers when used before Dial, so I don't think you can do this currently.š</div></div><div>...

....ru> wrote: > Hi, all > > Is there someway ability to insert custom Header to "SIP 486" message, > when HANGUP application is invoked? > > Our use case is to set that Header, when call-limit is reached, to analyze > elsewhere, but we do not want to set some custom causecode in HANGUP > application because this can confuse a calling equipment. > I only know of the SIPAddHeader application which lets you add headers when used before Dial, so I don't think you can do this currently. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Driv...

...t;> Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP >> application is invoked? >> >> Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but >> we do not want to set some custom causecode in HANGUP application because this can >> confuse a calling equipment. >> >> I only know of the SIPAddHeader application which lets you add headers when used before Dial, >> so I don't think you can do this currently. >> I think that Asterisk cannot handle th...

Hello all I have looked in Google for a fix for this issue but I have had no luck, so I am posting it here. Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting causecode to be single byte but was 35 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting samplerate to be 2 bytes long but was 40 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting samplerate to be 2 bytes long but was 40 Jun 10 03:42:17 WARNING[2961]: Chan_i...

Hi, We have quite a large setup working just fine most of the time. We have 60 outgoing lines on PRI and we never use all of these lines. But sometimes we get the "all circuits busy now" message, seemingly random. Sometimes we get it before the call even goes through to PSTN. Sometimes after 5 or 6 rings etc. It seems that the carrier is signalling something and asterisk always

I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060503/3cb7f966/attachment.htm

...;Hi, all<br /> <br /> Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked?<br /> <br /> Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.</blockquote><div>š</div><div>I only know of the SIPAddHeader application which lets you add headers when used before Dial, so I don't think you can do this currently.š</div></div><div>...

...74992a24..Call-ID: 422fd4375fe79a5 977e891870f5cc05b at 74.13.233.143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFE R, SUBSCRIBE, NOTIFY..Supported: replaces, timer..Content-Length: 0..X-Asterisk-HangupCause: Normal Clearing..X-Asterisk-Hangup CauseCode: 16.... # U 2009/07/15 22:30:17.186266 209.167.0.151:5060 -> 74.13.233.143:5060 ACK sip:123456 at 74.13.233.143 SIP/2.0..Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK00000000;rpo...

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown