similar to: Any way to get rid of X-Asterisk?

Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris

Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the

hello, i'm testing sending ISDN cause codes to customer pbx (test scenario for unallocated number) topology: PSTN-E1-AsteriskA-AsteriskB-SOMEPBX INVITE from SOMEPBX to PSTN AsteriskA sends to AsteriskB Status-Line: SIP/2.0 503 Service Unavailable X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 how can i resend HangupCauseCode from AsteriskB to

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar

Hello Everyone. Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls: Name/username Host Dyn Forcerport ACL Port Status Realtime 222/222 (Unspecified) D N A 0 Unmonitored Cached RT So when we DIAL 222 we get: WARNING[23103]: app_dial.c:2198 dial_exec_full:

I have two version 1.6 Asterisks running. One is a small hobbyist thing just at home, and the other is handling calls for several customers. On both, I have added the line exten => h,1,Set(CDR(hangupcause)=${HANGUPCAUSE}) to all relevant contexts. On my little hobbyist box this works perfectly; all calls have their hangupcauses recorded with cdr_adaptive_odbc and cdr_custom. On the

Thanks Dovid! Indeed looks a bug but regardless of this, this problem made me think that the HANGUPCAUSE could be used for this purpose with benefits. I couldn't find an explanation about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better

This was always possible in the past, however does not work in the current release. I believe this is a bug. To: asterisk-users at lists.digium.com From: cervajs at fpf.slu.cz Date: Fri, 9 Oct 2015 10:04:47 +0200 Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR search in archives save the records to another table like cdr_extended Dne

Hello list, Hope all doing well! I've been checking some cases when a Dial fails and dialplan execution continues to handle this. I am finding it a little confusing how we should handle the DIALSTATUS and the HANGUPCAUSE in this situation.... More specifically, I am facing a case in version 13.6.0 where I am getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me

Hi, I have the following code that operates when a channel is hung-up: [record-hangupcause]exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})exten => s,n,Return() Before the dial a hangup handler is registered: Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1) The routine is called and the variables are being set, however not on the channel's CDR which made the call. I believe this

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 -> 192.168.2.20:5080

Hi, I'm using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I've tested allow

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is