similar to: Channel in an unkown state

Displaying 20 results from an estimated 10000 matches similar to: "Channel in an unkown state"

2008 Oct 20
0
TDM410P with EC doesn't detect DTMF after being on for ~1 hour
Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [root at asterisk wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. -------------------- Use chan_dahdi.conf file to adjust the gains if needed. ( # =
2008 Apr 14
0
CallerID in NZ
Hi There, We have a Asterisk 1.4 box with a X100P card connected to a analog line with Caller ID serrvices enabled on it. When an incoming call appears we get the following in the log: -- Starting simple switch on 'Zap/1-1' -- Detecting post-CID distinctive ring [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... [Apr 15 10:38:07] NOTICE[7151]:
2010 Jan 05
5
CallerID on Indian PSTN is not working.
Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.
2010 Mar 12
0
Regarding - P-Asserted identity and Privacy - SOLVED
Hi All, I got this figured out, when the privacy is ON at the other end of the server and when we get the Invite message to the server connected to PRI's, just take the details from the invite message in the Dial plan and send the calls as anonymous: exten => _1NXXXXXXXXX,n,Set(PRIVACY=${SIP_HEADER(Privacy)}) exten => _1NXXXXXXXXX,n,ExecIf($["${PRIVACY}" =
2011 May 26
0
Dahdi channel stuck in "ringing" state
Hi, For some time now I have noticed that our RBS T1 (asterisk 1.4.35, Dahdi 2.3.0+2.3.0, TE410P) often has channels stuck in the state "Ringing", like this poor chap who got stuck on two calls in a row, apparently: [excerpt from "core show channels"] SIP/7157997-0000534b 7760308 at business:1 Ring Dial(Dahdi/g0/7760308) DAHDI/3-1 5130262 at from-pstn:1
2011 Jan 21
1
Unable to receive calls (inbound)
Hello all. I have installed AsteriskNow 1.7.1 with all updates. I'm able to make outbound calls without any problem (the external calls are made via an analog line, and the receiver see the CID). However I'm unable to forward incoming calls to the destination I want. What happens is when I make an internal call I ear a "bye". Bellow is the log of the internal call: --
2011 Jan 21
1
Inbound routes
Hello all. I have installed AsteriskNow 1.7.1-64bits with freePBX. The system has 1 DAHDi card with 2 analog FXO ports (to pstn) and 1 FXS port connected to a FAX machine. I want the every call received on port FXO-2 to be redirected to the FAX machine. So, what I configured was that every call with DID 12345 (let's suppose that 12345 is the DID connected to the FXO-2)to be redirected to the
2009 May 14
3
how to avoid call waiting? Or check DIALSTATUS before Dial()?
I have two internal analogue extensions off a TDM400P. If the first is busy, I'd like to ring the second. So: [incoming] exten =>s,1,Answer() exten =>s,n,Dial(${mainline},60) exten =>s,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Dial(${secondline},30)) But it doesn't work because * first tries Call Waiting on the main line. Here I dial out: -- Starting
2013 Jan 24
2
Asterisk 11 / Missing Application SetCallerPres
Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]
2010 Mar 26
1
SIP/2.0 403 Forbidden
hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
2009 Oct 26
1
DAHDI not detecting RINGING Status on the Channel
I am using an 8 port tdm card and also I implemented a dialer using a .call file generator. As you know on the .call you specify the channel to call and then the contex/extension/priority to let dial plan continue when the call is bridge. My actual problem is that when the call process starts, asterisk (DAHDI) sets the channel as answered when the truth is that on the other side the channel has
2010 Mar 26
1
send a call from A to B use sip trunk prablem
i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
2009 Oct 09
0
calls ansowered for 1 second or less
Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, it?s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : <vendor> ----SIP--? <Asterisk> ?----IAX2---? <Asterisk with Dhadi channels> Here: -- Executing [966505103150 at from-internal:1]
2010 Jun 21
1
How to tell if a dropped call is my fault
I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP
2011 Jan 26
0
Really wacky problem with internal extensions.
We have an Asterisk server acting as a hosted PBX system for many clients, and we're going through an upgrade to Asterisk 1.6 by moving our most important (and complicated) clients one at a time. But we're having a problem with one customer that I really can't explain. I can place calls directly to one phone at the customer's location (they also have an IVR that asks for an
2010 Jan 19
0
Detecting incoming faxes and forwarding to phone fax machine
I'm having a problem receiving incoming faxes and I'm hoping someone here can help me out. My system is a PBX in a Flash with one dahdi card for my incoming analog lines and another dahdi card for my analog devices (fax and modem). My dahdi-channels.conf file looks like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 23 14:56:24 2009 ; If you edit this file and execute
2009 Feb 23
1
Inbound call to IVR drops after 21 seconds?
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]
2009 Mar 17
0
Weird issue with outbound calls and MOH
Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to
2013 Sep 25
2
users can not hear the audio playback sometimes
Hello everyone, I am facing a strange problem on my asterisk box (using isdn lines with pri card installed on it). Normal incoming/outgoing calls are working perfectly fine. When a user dial a wrong/out-of-service number they don't hear back any such message like "The number is wrong or user is switched off" in some cases, and it's just a silence for the user. Now while