similar to: Asterisk dialing next extension only if first is busy?

What am I doing wrong? Goif dialstatus: busy CONGESTION not working. exten => _7NXXXXXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr) exten => _7NXXXXXX,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2) exten => _7NXXXXXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr) exten => _7NXXXXXX,n,Hangup() When I try to

The "error" I was talking about was in your log: "...== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'..." The call terminated here in a error which prevented the dialplan from continuing. Something there is broken, my recommendation is to check you registrations first inside asterisk: > sip show peers Something wasn't

I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done

> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote: > > On 1/2/24 15:13, asterisk at phreaknet.org wrote: >>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: >>> I'm using asterisk-16.30.1 >>> >>> When I try to call another asterisk server over IAX I get a busy signal, >>> >>> chan_iax2.c:4739 __auto_congest:

I was wondering if it is possible (in 1.2.x) to get the SIP response code back after doing Dial(). Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and some are NOANSWER, but I want to know the SIP response code, so I could return the right tones to the user, not just a congestion tone for every fault. Anyone know a way to find out that information, so I want the

G'day. I am wanting to find out how my SIP service is performing with Asterisk, especially jitter and dropped packets. I can get an overview of that using the 'rtcp stats' function at the console, but is there any way to get those logged to a file or some other permanent record? Nothing in logger.conf seems applicable, save perhaps directing verbose messages somewhere, but it

On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: > I'm using asterisk-16.30.1 > > When I try to call another asterisk server over IAX I get a busy signal, > > chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow > response >     -- IAX2/192.168.143.1:4569-656 is circuit-busy > > Asterisk-16.16 is working normally, no congestion error. There is not

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed

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

I've seen several examples that use extensions such as; s-BUSY s-NOANSWER etc... It's more or less evident what they do, but I've searched for some FORMAL documentation everywhere and have found nothing. Do they work for anything else than "s-"? (I think I've seen other examples, but can't find them now) Are they standard in any way? What are the allowed values

Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

G'Day All, Newbie here. How can I go about troubleshooting a fast busy when I dial my the phone number on my * server? Thanks. Ferg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041012/a669d795/attachment.htm

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

G'day. I have a UPS here that is driven by the Fenton UPS driver, and which requests that I report the ID string to the NUT developers: root@anu:~# upsdrvctl start smk800 Network UPS Tools - UPS driver controller 2.0.4 Network UPS Tools - Fenton UPS driver 1.22 (2.0.4) Unknown ups - please report this ID string: #M+H SMK-800 2000 V6.2 Detected Unknown MK-800 on /dev/ttyS0 The

I've not seen an answer to this in any forum. I make a call through Asterisk, with a VOIP phone, doesn't matter which. The call gets made, I leave a voicemail, or complete the call in some manner, and the other side hangs up. I hear a busy signal on the phone on my end. If I have an extension that looks like this, after the hangup() is executed, my phone gives busy signals until I

Hi, I was wonder is there any way to custom the message on the call busy or no answer I actually get the error code from asterisk server on busy or no answer. Can I custom the text message or custom the message to sound ? Anyone have any idea could u please share me ? Thank, Thyda -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current 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

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from