Elastix 1.6-12 and OpenVox A400E

ScottMcKeown

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#1
Hi All,

I'm fairly new to the Asterisk & Elastix platform but I'm managing to find my way throug with some help from past posts and the wonderful "Elastix Without Tears" (thank you Ben).

OK I'll try and give as much information as possible but if you need anything else please just ask, also please be paitent with me if I don't fully understand I may have to go and pay another visit to the PDF of the forums first. ;)

I have got a fully working SIP Trunk from a UK Telco working fine with so far 6 users all using the system both inbound and outbound calls are fine. What I'm working on now is the backup PSTN(POTS) line that I have been asked to put into the system just incase we loose our connection to the SIP Gateway or the internet dies :woohoo:

I'm using Elastix 1.6-12 which has been fully updated via the freePBX web interface along with an OpenVox A400E card with 2 Port FXS enabled.

I have now read through the "Elastix Without Tears" manual about 3 times and I'm still none the wiser as to how I can get this to work.

The Elastix "Hardware Detection" page shows

The problem is that I have created the Trunk and added one Queue with just a voice recording to prove that it works but I get nothing.
The PSTN Trunk

I can not for the life of me find any of the Zaptel Config files but I can find the DAHDI files but they look nothing like the ones in the PDF that I have.

If anyone can please give me a nudge in the right direction I would be most greatful.


~Yours,
Scott
 

Joe.Yung

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#2
Hello Scott,
Read this link for more:http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/
For the PSTN Trunk settings.
Here is a simple sample, and hoping it will help you a little. (the settings of OpenVox A400P is the same as A400E)
http://downloads.openvox.cn/pub/misc/In ... 00p_En.pdf

Regards!
Joe
 

dicko

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#3
Joe.Yung:

You are totally and essentially wrong here, You need to download and actually read the current version of "Elastix Without Tears" (May 8th 2009), we await your retraction of your factually erroneous post after you do your belated "due diligence" , and also for you and your company to significantly improve your support of the poor Elastix users who are suckered into going cheap and essentially unsupported here with your products.


I quote Belief Mo

"We certainly recommend this project for PBX projects"

Yet you have been absent in any useful reciprocation to Elastix for two years. (ask the esteemed Fernando Villares for his opinion)

My personal experience with your stuff and my personal rebuttal to Belief is:

"I can not in any way recommend OpenVox products if you want a seamless integration of your hardware into Elastix, they show zero commitment to Elastix, or the understanding of it's ethos"

regards


dicko
 

james.zhu

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#4
hi:
thank you, dicko for your input. openvox's guys are here to support
the users who are using Elastix with OpenVox. we are not politician and not
play a political games. It is very true, digium and sangoma are very good in term of ..., personally, i respect them and appreciate them. we are tech guys ONLY. to this case, we have to clarify:
1) does the card work?
2) does the user sets it in a right way?
3) has he asked/contacted OpenVox before posting this problem?
ScottMcKeown, please contact Joe or me and send ssh to us.
 

OpenVox

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#5
Hi Dicko,

First of all, as a user of Elastix, we really appreicate your great contributions to the elastix community and the open source community. You do a great jot to help other poeple. You deserve a clap!

Second,the open source community is built by all of us, not just by one single person or company! Everybody has the right to play a role here if he wishes. So does OpenVox. OpenVox was founded in 2002 and almost 8 years into the open source community. We try every possible way to contribute more to the open source community. And we do.Thanks for our clients who supprt OpenVox all the time, we grow from nobody to somebody! And now you know OpenVox and people know OpenVox all around even if you do not like us, but we are just here! No company acts perfect! Neither sangoma nor Digium! That's the reason why we need to grow. Because we know that we are not perfect! And thus we have our technical support here to help. We want to show our customers that we are always behind to support our products, solve the problems so that we can make sure these problems are not going to happen next time! Did you find any topic that is not answered and followed by our technies if posted in OpenVox forum when it's hardware issues? Did you find problems solved by our technies in other brand's forum? Why we do this? Because it an open source community and we are not here to ask for what open source community can give us but to show the people what we can do for the community!

And it's not the only thing we do to help our customers: we provide THREE-MONTH no questions asked return policy!If you do not satisfied, just return it! Are you sure you know OpenVox well?

We are sorry if we made a mistake of the release for Ben's book.If Scott did not tell us, we may be hard to know the right veriosn. The lastest veriosn of the book is updated on April 3, 2010. downloadable at http://elastixconnection.com/downloads/ ... _tears.pdf

For the comments on Elastix's homepage, do you know when it's posted? It was in 2007 when Elstix was a kid in the community and we provide hardware support to help the project and don't you think it's great to recommend Elastix for PBX projects when it starts and tribox is all arround that time? Did you notice that OpenVox is listed in the first place for over two years? Elastix and OpenVox have their own way to cooperate. How do you know that we are doing nothing to the Elastix community even if you know nothing about us?

I repsect your own idea if you do have experience in using our products and it turns out to be that you don't like it and we are sorry that our products can not satisfy all your needs. You've the right to choose other brands.

But Dicko,you don't have to act this way to show us your favors. Live and let live.

Best Regards,
Belief Mo
OpenVox
 

ScottMcKeown

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#6
Hi Guys,

First off thank you for taking the time to help me with this issue, but lets now consider our opinions as voices and move forward with the issue in hand.

I must confess that I was reading an outdated copy of the PDF
 

ScottMcKeown

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#7
Hi Guys,

Sorry for posting twice in one day but could the following be of any help?

dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.2.0.2
ACPI: PCI Interrupt 0000:02:00.0[A] -> GSI 16 (level, low) -> IRQ 169
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Not installed
Module 2: Not installed
Module 3: Not installed
Found a Wildcard TDM: Wildcard TDM400P REV E/F (1 modules)
dahdi_transcode: Loaded.
INFO-xpp: revision trunk-r6963 MAX_XPDS=64 (8*8)
INFO-xpp: FEATURE: with BRISTUFF support
INFO-xpp: FEATURE: with PROTOCOL_DEBUG
INFO-xpp: FEATURE: with sync_tick() from DAHDI
INFO-xpp_usb: revision trunk-r6963
usbcore: registered new driver xpp_usb
Unified AP4XX PCI Card Driver
DAHDI Dynamic Span support LOADED
rxt1: no version for "dahdi_hdlc_putbuf" found: kernel tainted.
dahdi_echocan_oslec: Registered echo canceler 'OSLEC'
dahdi: Registered tone zone 0 (United States / North America)
All TDMoE multiframe span groups are active.

I only have one line enabled and I'm passing my BT telephone line into this port but as I'm based in the United Kingdom could it be due to the line:

dahdi: Registered tone zone 0 (United States / North America)

If so how do I change this?


~Yours,
Scott
 

dicko

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#8
Belief:

I posted in response to an incorrect statement from your "techie", when he posted originally that "Eastix Without Tears" was based on ZAP, that has not been true for over a year. I expected a retraction, we got a redaction (he erased it from his post)

I added my opinion of your product, this opinion has developed over the years, when I encounter your hardware (and other digium clones), usually by inheritance, I have noticed that replacing them with Digium hardware usually resolves all problems with noise, echo and level in the analog interfaces, this might be a coincidence, but then again it might not be.

Do you in any way deny that your designs are largely "reference design" rip-offs of someone else's original work with no actual substantive changes hardware or software advanced by yourself apart from "capacitors" and vendor ID's, do they fully follow the digium dahdi directives?

If I am wrong, then I suggest that is up to you to submit your "improvements", accept peer review, and hopefully for you, become mainstream with digium/asterisk , no?


It is of course just my humble opinion yet I own it and stand by my statement.


I refer ScottMcKeown's opermode question to:


http://www.voip-info.org/wiki/view/TDM400P

dicko
 

ScottMcKeown

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#9
dicko,

Once more thank you for your time and help.

I will have a read and if I have any further issues you can rest assured that I will be back.

However, I may have been a little bit of an idiot as I have just noticed that the Card I have installed only has an FXS module installed which if my basic level of POTS is correct this is meant for connection a phone/fax/modem into the system and not for connecting a POTS line to.

Or do I have that wrong? I'm sorry but I have now read and googled that much stuff I'm getting myself even more confused.


~Yours,
Scott
 

dicko

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#10
Indeed FXS is for a phone, and yes you will need an FXO for a trunk connection. And yes it is confusing that FXS hardware uses FXO signaling. (because it looks like an FXO from the outside), but hang in it will all start to become clear.

good luck

dicko
 

ScottMcKeown

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#11
Once again thank you for your help dicko.

Well my extension module has just arrived and I'm off to play again I'm sure I'll have more questions later but thats for a different thread I think.

Thank,

Scott
 

ScottMcKeown

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#12
Well this is starting to get more than just a little frustrating...

... I've followed the guides and even tried installing Elastix 2.0 RC2 and still can not get this card to function the way I want it too.

I'm now getting the following in my logs when I make a call to the number on my PSTN line:
[May 6 18:04:18] NOTICE[7016] chan_dahdi.c: Alarm cleared on channel 2
[May 6 18:04:19] WARNING[7016] chan_dahdi.c: Detected alarm on channel 2: Red Alarm
[May 6 18:04:39] NOTICE[7016] chan_dahdi.c: Alarm cleared on channel 2
[May 6 18:04:40] WARNING[7016] chan_dahdi.c: Detected alarm on channel 2: Red Alarm
[May 6 18:04:42] NOTICE[7016] chan_dahdi.c: Alarm cleared on channel 2
[May 6 18:04:43] WARNING[7016] chan_dahdi.c: Detected alarm on channel 2: Red Alarm
[May 6 18:05:21] NOTICE[7016] chan_dahdi.c: Alarm cleared on channel 2
[May 6 18:05:22] WARNING[7016] chan_dahdi.c: Detected alarm on channel 2: Red Alarm

I have googled this and I get 2 results back which are less than helpfull - I have also e-Mailed OpenVox support but so far after 3 days nothing back.

Can anyone out there please give me some help on trying to work out what this means.


Thank you very much,


Scott
 

rollinsolo

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#13
Scott did you by chance actually follow the tutorial on the openvox site and download dahdi-complete and recompile it, because if you did, it will mess things up for you. I notice that when you try to recompile new drivers on Elastix, some things seem to go wrong.
so: 1. if you did follow the openvox wiki and downloaded dahdi again, go ahead and reinstall 1.6, do not touch dahdi, just leave your card in there, then click the hardware detection, and then go to dahdi-channels.conf and make sure you have the right amount of channels there. Save it and reboot. Then try the calls before you manipulate anything else in system.conf, chan-dahdi.conf, and dahdi-channels.conf. Such as US to UK (if that is what you change it to lol sorry I dont know)
 

ScottMcKeown

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#14
Hi rollinsolo,

Thank you for the reply, and sorry its taken so long for me to get back to you all (had a few days off).

Originally, yes I did follow the guide from OpenVox about installing the updated dahdi but that did not work so I scrubbed it and started again thinking that I had messed it up.

I now have a fresh install that I have left alone waiting for some help from OpenVox which seems to be none existant at the moment even after repeated e-Mails (I've not even had a bounce so I know they were delivered).

I think what I'll have to do is start again and keep posting my steps in here.

Can I ask are you using an OpenVox A400 card?

I know your in the States but if I know someone that has got the card working then I maybe able to ask for a comparison on config files to make sure that I'm at least looking in the right place.


~Thanks,
Scott
 

OpenVox

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#15
Hi Scott
 

ScottMcKeown

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#16
Hello OpenVox,

These are the e-Mail addresses that I have sent the e-Mails too.

I'm sorry to say that it looks like I was a bit to house proud and cleaned out me sent items folder a few days ago so I can't now even tell you when I sent the e-Mails.

However, I have just sent another e-Mail to both addresses now.


~Yours,
Scott
 

OpenVox

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#17
Hi Scott,

Ok, we will check the email and see how we can help you to get the problem solved.

Thanks.

Best Regards,
Belief Mo
OpenVox
 

ScottMcKeown

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#18
Thank you

~Scott
 

ScottMcKeown

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#19
Hi All,

Well heres an update for anyone that maybe interested or is trying to use BT with an OpenVox Card.

We spent about 4 hours via MSN messanger yesterday trying to get this to work and now we are nearly there.

My /etc/asterisk/dahdi-channels.conf now looks like this:
; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
;;; line="1 WCTDM/4/0 FXOKS"
signalling=fxo_ks
callerid="Channel 1" <4001>
mailbox=4001
group=5
context=from-internal
channel => 1
callerid=
mailbox=
group=
context=default

;;; line="2 WCTDM/4/1 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
;context=from-zaptel
channel => 2
callerid=
group=
context=default

And my chan_dahdi.conf looks like this:
[trunkgroups]

[channels]
loadzone=uk
defaultzone=uk
usecallerid=yes
ukcallerid=yes
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity ; Added for UK CLI detection
sendcalleridafter=2
callerid=asreceived ; propagate the CID received from BT.

context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
;faxdetect=incoming
;echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=3

immediate=yes

#include dahdi-channels.conf
#include chan_dahdi_additional.conf


However, we also discovered that I needed to get BT to change something called the "Disconnect Clear Time" changed from 100 to 800 which was done in about 10 minutes once I got hold of BT.

This now leaves me with the following messages in the logs:
[May 14 12:19:01] VERBOSE[5761] logger.c: == Starting post polarity CID detection on channel 2
[May 14 12:19:01] DEBUG[5761] dsp.c: dsp busy pattern set to 0,0
[May 14 12:19:01] VERBOSE[6240] logger.c: -- Starting simple switch on 'DAHDI/2-1'
[May 14 12:19:01] NOTICE[6240] chan_dahdi.c: Got event 17 (Polarity Reversal)...
[May 14 12:19:01] DEBUG[6240] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 2, state 9
[May 14 12:19:01] DEBUG[6240] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 2, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1769344972
[May 14 12:19:02] DEBUG[6240] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 2, state 9
[May 14 12:19:02] DEBUG[6240] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 2, state 9
[May 14 12:19:02] DEBUG[6240] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 2, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1769344780
[May 14 12:19:03] WARNING[6240] chan_dahdi.c: CID timed out waiting for ring. Exiting simple switch
[May 14 12:19:03] VERBOSE[6240] logger.c: -- Hungup 'DAHDI/2-1'
[May 14 12:19:04] VERBOSE[5761] logger.c: == Starting post polarity CID detection on channel 2
[May 14 12:19:04] DEBUG[5761] dsp.c: dsp busy pattern set to 0,0
[May 14 12:19:04] VERBOSE[6241] logger.c: -- Starting simple switch on 'DAHDI/2-1'
[May 14 12:19:04] NOTICE[6241] chan_dahdi.c: Got event 17 (Polarity Reversal)...
[May 14 12:19:04] DEBUG[6241] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 2, state 9
[May 14 12:19:04] DEBUG[6241] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 2, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1769341932
[May 14 12:19:05] DEBUG[6241] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 2, state 9
[May 14 12:19:05] DEBUG[6241] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 2, state 9
[May 14 12:19:05] DEBUG[6241] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 2, state 9, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1769341852
[May 14 12:19:06] WARNING[6241] chan_dahdi.c: CID timed out waiting for ring. Exiting simple switch
[May 14 12:19:06] VERBOSE[6241] logger.c: -- Hungup 'DAHDI/2-1'
[May 14 12:19:06] VERBOSE[5761] logger.c: == Starting post polarity CID detection on channel 2
[May 14 12:19:06] DEBUG[5761] dsp.c: dsp busy pattern set to 0,0
[May 14 12:19:06] VERBOSE[6242] logger.c: -- Starting simple switch on 'DAHDI/2-1'
[May 14 12:19:07] NOTICE[6242] chan_dahdi.c: Got event 17 (Polarity Reversal)...
[May 14 12:19:09] WARNING[6242] chan_dahdi.c: CID timed out waiting for ring. Exiting simple switch
[May 14 12:19:09] VERBOSE[6242] logger.c: -- Hungup 'DAHDI/2-1'

I have checked with BT and the line is working as it should do but I have found that due to the age of the line "Caller ID" was not enabled, so I'm now waiting for this to be activated and should be done within the next 24/48 hours.

So I've now set up the PSTN line to come straight in and play a recorded message and then force a hangup which should also enable the line to be kept clear for the next caller.

I'll post again once Caller ID is enabled.


~Scott
 

ScottMcKeown

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#20
OK well BT have informed me that CID is now enabled and it sort of works now. When I make a call to the PSTN/POTS line I get the following from the CLI:

== Starting post polarity CID detection on channel 2
-- Starting simple switch on 'DAHDI/2-1'
-- Executing [s@from-zaptel:1] NoOp("DAHDI/2-1", "Entering from-zaptel with DID == ") in new stack
-- Executing [s@from-zaptel:2] Ringing("DAHDI/2-1", "") in new stack
-- Executing [s@from-zaptel:3] Set("DAHDI/2-1", "DID=s") in new stack
-- Executing [s@from-zaptel:4] NoOp("DAHDI/2-1", "DID is now s") in new stack
-- Executing [s@from-zaptel:5] GotoIf("DAHDI/2-1", "1?zapok:notzap") in new stack
-- Goto (from-zaptel,s,8)
-- Executing [s@from-zaptel:8] NoOp("DAHDI/2-1", "Is a Zaptel Channel") in new stack
-- Executing [s@from-zaptel:9] Set("DAHDI/2-1", "CHAN=2-1") in new stack
-- Executing [s@from-zaptel:10] Set("DAHDI/2-1", "CHAN=2") in new stack
-- Executing [s@from-zaptel:11] Macro("DAHDI/2-1", "from-zaptel-2|s|1") in new stack
-- Executing [s@macro-from-zaptel-2:1] NoOp("DAHDI/2-1", "Entering macro-from-zaptel-2 with DID = s and setting to: 01329230498") in new stack [May 17 12:34:02] DEBUG[5849]: app_macro.c:379 _macro_exec: Executed application: Noop
-- Executing [s@macro-from-zaptel-2:2] Set("DAHDI/2-1", "__FROM_DID=01329230498") in new stack [May 17 12:34:02] DEBUG[5849]: app_macro.c:379 _macro_exec: Executed application: Set
-- Executing [s@macro-from-zaptel-2:3] Goto("DAHDI/2-1", "from-trunk|01329230498|1") in new stack
-- Goto (from-trunk,01329230498,1)
[May 17 12:34:02] DEBUG[5849]: app_macro.c:379 _macro_exec: Executed application: Goto
== Channel 'DAHDI/2-1' jumping out of macro 'from-zaptel-2'
-- Executing [01329230498@from-trunk:1] Set("DAHDI/2-1", "__FROM_DID=01329230498") in new stack
-- Executing [01329230498@from-trunk:2] Gosub("DAHDI/2-1", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("DAHDI/2-1", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("DAHDI/2-1", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Set("DAHDI/2-1", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:4] Return("DAHDI/2-1", "") in new stack
-- Executing [01329230498@from-trunk:3] ExecIf("DAHDI/2-1", "1 |Set|CALLERID(name)=") in new stack
-- Executing [01329230498@from-trunk:4] Ringing("DAHDI/2-1", "") in new stack
-- Executing [01329230498@from-trunk:5] Set("DAHDI/2-1", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [01329230498@from-trunk:6] SetCallerPres("DAHDI/2-1", "allowed_not_screened") in new stack
-- Executing [01329230498@from-trunk:7] Goto("DAHDI/2-1", "app-announcement-1|s|1") in new stack
-- Goto (app-announcement-1,s,1)
-- Executing [s@app-announcement-1:1] GotoIf("DAHDI/2-1", "0?begin") in new stack
-- Executing [s@app-announcement-1:2] Answer("DAHDI/2-1", "") in new stack [May 17 12:34:02] DEBUG[5849]: chan_dahdi.c:3699 dahdi_answer: Took DAHDI/2-1 off hook [May 17 12:34:02] DEBUG[5849]: chan_dahdi.c:2278 dahdi_train_ec: Engaged echo training on channel 2
-- Executing [s@app-announcement-1:3] Wait("DAHDI/2-1", "1") in new stack [May 17 12:34:02] DEBUG[5849]: chan_dahdi.c:5473 dahdi_handle_event: Polarity Reversal event occured - DEBUG 1: channel 2, state 6, pol= 1, aonp= 1, honp= 1, pdelay= 600, tv= 57 [May 17 12:34:02] DEBUG[5849]: chan_dahdi.c:5480 dahdi_handle_event: Polarity Reversal detected but NOT hanging up (too close to answer event) on channel 2, state 6 [May 17 12:34:02] DEBUG[5849]: chan_dahdi.c:5487 dahdi_handle_event: Polarity Reversal event occured - DEBUG 2: channel 2, state 6, pol= 1, aonp= 1, honp= 1, pdelay= 600, tv= 57
-- Executing [s@app-announcement-1:4] NoOp("DAHDI/2-1", "Playing announcement hangup") in new stack
-- Executing [s@app-announcement-1:5] Playback("DAHDI/2-1", "busy-hangovers|noanswer") in new stack
-- <DAHDI/2-1> Playing 'busy-hangovers' (language 'en')
-- Executing [s@app-announcement-1:6] Goto("DAHDI/2-1", "app-blackhole|hangup|1") in new stack
-- Goto (app-blackhole,hangup,1)
-- Executing [hangup@app-blackhole:1] NoOp("DAHDI/2-1", "Blackhole Dest: Hangup") in new stack
-- Executing [hangup@app-blackhole:2] Hangup("DAHDI/2-1", "") in new stack
== Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'

Which is what should be happening. The call should come in and get passed to an announcement (one of the system pre recorded ones) and then hang up the call to allow another call to be made. However, I'm not hearing anything on the phone and the call does not get hungup but just keeps ringing, other than that the system seems to be following the config.

Anyone got any small idea on what I could have missed?


~Ta
Scott
 

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