similar to: IVR Loop on invalid input

Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in

Does anyone know why? ThePBX*CLI> -- Executing [310-456-7890 at from-trunk:1] Set("SIP/202.101.202.101-b763ce60", "__FROM_DID=310-456-7890") in new stack -- Executing [310-456-7890 at from-trunk:2] ExecIf("SIP/202.101.202.101-b763ce60", "1 |Set|CALLERID(name)=310-456-0987") in new stack -- Executing [310-456-7890 at from-trunk:3]

Having some issues with getting sound from a call. I have 4 systems. 3 main systems which handle calls for our 3 locations. The 4th system is the central voice mail system. When an inbound call gets passed to someones voice mail its done with an IAX2 connection. The same happens after hours when we have our night mode set. If you dial the main number after hours you are passed straight to the

Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

The following code using OpenMP pragmas , when compiled with gcc 4.5 + LLVM 2.8 + DragonEgg 2.8 and ran, produces segmentation fault. //----------------------------------------------------------- #define LOOPCOUNT 10000 int main() { int bit_and = 1; int logics[LOOPCOUNT]; int i; for (i = 0; i < LOOPCOUNT; ++i) { logics[i] = 1; } #pragma omp parallel for schedule(dynamic,1)

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Hi all, I have just setup a trixbox system and I am implementing hylafax/iaxmodem solution for the faxing. When i send a fax to it by phoning in listening to the IVR and manually pressing start to initate the fax, the call gets picked up correctly as a fax and everything works well. When I try sending a fax by entering the phone number and pressing start to initiate dialing it sounds like

Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile

Hi Everyone, I have done some php coding to come up with my own FollowME module for FreePBX. The need for this has some security considerations behind it. This is what my code does at core: $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing, pre_ring) VALUES

Hi Guys, I am making another module for Voicemail. I have three fields in a POST form that have to be connected together to make it a single 10 digit number but there is something wrong in my syntax probably. $npaa = "('$_POST[anpa]')"; $nxxa = "('$_POST[anxx]')"; $blocka = "('$_POST[ablock]')"; *$grplist = $npaa.$nxxa.$blocka;*

On 04 Aug 2014, at 09:27, Reid Kleckner wrote: > On Sat, Aug 2, 2014 at 7:51 AM, Jonas Maebe > <jonas.maebe at elis.ugent.be> > wrote: > >> On 01/08/14 19:37, Reid Kleckner wrote: >> >>> What happens if you drop appending linkage? I think it will just >>> work, >>> since you are already using a custom section, which will ensure

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

On 11/02/2011 11:17 AM, Marcello Maggioni wrote: > Mmm I found out a very strange behavior (to me) of the SCEV analysis > of the loop bound of the external loop I posted. > When in ScopDetection it gets the SCEV of the external loop bound in > the "isValidLoop()" function with: > const SCEV *LoopCount = SE->getBackedgeTakenCount(L); > > It returns a

On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as

OK, this is how I thought it's supposed to work. It just confounded me why the book would say the Playback() and Background() syntax were the same, then in the very next paragraph give an example that belied that claim. On 6/17/2023 1:46 PM, Doug Lytle wrote: > On 6/17/23 08:47, Steve Matzura wrote: >> >> Both Background()  and WaitExten()  allow the caller to enter DTMF

I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS. The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log : ==== Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command' Apr 30

Hi Tobias. I worked on enabling Polly accepting non affine memory accesses and I produced a patch. I saw that there were a lot of updates in Polly recently, so I had to redo a lot of the work I did and that slowed me quite a bit. I tested the patch on some programs and it seems to work and not to break anything, but you know the topic much better than me, so if you find something wrong please

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: