similar to: Asterisk Hangup Issue in Ringing State with Incoming call

Hi, I am trying to build asterisk with xtensa compiler on embedded platform. I am trying to integrate my driver code to asterisk. For this tying to call driver code IOCTLs from chan_dahdi instead of dahdi IOCTLs. While compiling asterisk with xtensa, Observed chan_dahdi is not compiling [chan_dahdi.so is not creating].But with default settings [/usrc/src] it was creating. Please let me

Hi All, I am running asterisk on Linux machine and trying to use confbridge application. Please have a look at Conf files. sip.conf ====== [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow = all allow=ulaw allow=alaw defaultexpiry=100 [5001] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all

I've got a strange issue where I get no ringing on outbound PRI calls using the TE410P span 1. The call actually goes out and works, you just hear no ringing. The quality and features on the call are pristine, no cracks pops or any weirdness like that. If I specify the 'r' option on the Dial() string, I get a half of a ring, sometimes a full ring as soon as the dial completes.

Hi, I am trying to make external sip calls by using asterisk. Please provide information regarding sip trunk configuration in conf files. Setup is as below, * * *Case A:* Register two soft phones [X-lite] with 1000 and 10001 numbers to asterisk PBX [running in 192.168.1.11] and able to make calls in between. Sip.conf ====== [general] context=default bindport=5060

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi!. I am having problems with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I

Hello, I'm having a bit of trouble getting asterisk + tdm400p + rhino channel bank set up just right. The problem is a single ring on a telephone connected to the channel bank after the line is hung up. I've seen a couple messages similar to this, but I haven't seen any resolutions to the problem. Using the tdm400p card, with the first port connected to a 24-port fxs rhino channel

Hi, I have little problem and it is so embracing when u r talking to some one and line get hang-up. When some one calls from out of state or out of country my calls gets randomly hang-up with in few seconds and it happens with most of the calls. It's happening randomly I got few calls, which worked. I have also observer that with quite few call I can't hear the person and person can

Hi, I've set an Asterisk 11 box with a TDM400 board and Dahdi 2.7.0.1. I've connected an FXS port to an FXO one and issued a couple of channel originate command to measure the duration Asterisk/Dahdi needs to detect a dahdi call is coming in. Basically, using EPOCH variable, I'm reading a 2 or 3s duration with the followinf AEL2 dialplan: context remote { s => { if

Bug in "SOM" package in R. This simple code withot for loop runs:-------simple.r-------- library(som) A <- as.matrix(read.table("C:/code/h.txt", sep="\t", header=TRUE, row.names=1)) print(som(A,1,2)$visual) print(som(A,1,3)$visual) ---------------------------------------- The same code if inserted in for loop generates "Assertion failure" in

Hi all, I bought a Sangoma A200 card from an online supplier and explained exactly what I wanted, 3 incoming phone lines to PBX and a life line (some where to connect a standard BT phone to the PBX incase the power goes, making the BT phone ring). I was told to order 1 x FXS module (2 FXS ports) 2 x FXO modules (4 FXO ports) However being a complete noob, I have connected the 3 lines to the

Hello, Thanks for the clarifications plugin { quota = maildir:User quota quota_rule = *:storage=1G quota_grace = 10%% # 10% is the default quota_status_success = DUNNO quota_status_nouser = DUNNO quota_status_overquota = "552 5.2.2 Mailbox is full" } The above configurations is my default for all users. but my mail system is using mbox format

Hi All, does the Asterisk's 'Dial' command have some hooks to execute commands as soon as the 'ringing' signal is received ? For example: can a call be dropped 1 second after the called party's phone started to ring ? I'm using version 1.4. Thanks for supporting, have a nice day. Mike

I ran into a couple of odd problems with a new Asterisk box I have set up... I have a T100P connected to a channelized T-1 (24ch) to the PSTN. The T-1 is set up properly so far as I know and seems to work perfectly with one exception: When a call comes in, asterisk starts a 'simple switch' for it, goes through my 's' extension progression until it gets to the first Playback

Dear All I have this setup [asterisk]----[mediant2000]-------E1 Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip

I have 3 phones, SIP/3, SIP/6 and SIP/9 SIP/3 subscribes on hint on SIP/9 Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it is steady red. That's correct. But when 9 hangs up the hint goes to "InUse&Ringing", the light on 3 is still flashing. It keeps flashing until somebody calls 9 and hangs up again. -- Executing [9 at local:1]

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

I have had to install a TDM800 in a site, as the telco has held off installing ISDN indefinitely.. It's all fine except for the fact that it takes ages to hang up the line (6 or more rings), and sometimes doesn't even bother. This is only on incoming calls - outgoing calls work perfectly. Is there any good tricks for a fast and accurate hangup detect in this situation?

Hi, I was about to send a question to the list about this when I figured out the solution. I thought it might be useful to store in the archives. I use a X100P and a TDM400P, using current code from CVS. When I dial in via the X100 I get -- Starting simple switch on 'Zap/1-1' right away in the CLI. However, it takes (took) * a few seconds to continue patching the call through to

Hi All, I wasn't too sure how to word the subject, so I apologise for that. Anyway, I've got two X100P cards here accepting calls. Basically in Australia our ring cadence is 400ms on, 200ms off, 400ms on, 2000ms off, repeat. What I've noticed is that it takes about 8 seconds after the caller has hungup for the internal phones to stop ringing, whereas in Australia the maximum time

Hi, I've noticed that if a phone goes UNREACHABLE while it is Ringing, when the phone comes back, Asterisk will not clear the channel that was created, so it still thinks it is in the Ringing state. The only way to clear this is to do a soft hangup on the SIP channel or to restart Asterisk. Unfortunately these issues are very hard to automatically track down and clear and it seems like if a