similar to: Sending a hook flash to a DAHDI channel

HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable

-- Attempting call on DAHDI/1wwwwww for smvoice_callprogress at smvoice-dialout:1 (Retry 1) [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request: Unknown option 'w' in '1wwwwww' [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) Does DAHDI not know about the W ??? I think zaptel used

I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-9999) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any

Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as of a week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules and 1 VPMADT032 Module, hooked up to 5 analog lines. I get the error message referenced in the subject in my dmesg output everytime I load / reload DAHDI using the command "system dahdi start/restart". When I make an outbound call over

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asteriskcdrb"* table and it's pretty much useless in this case as it only logs the duration and

Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways,

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is

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

Hi all! I have installed a quite old Asterisk 1.6.2.0-rc2 with latest DAHDI on Ubuntu 9.10 from repository. It is working now but mysql logging is very strange. All calls have logged in mysql cdr table, which is fine, but disposition is 'NO ANSWER' even if I had talked on phone. Duration is correct but billsec is zero. Any idea why? Unfortunately I cannot upgrade to newer version because

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: --

Is anybody else using the following combination: * a TE410P card (wct4xxp driver) * a BT ISDN connection * DAHDI 2.3.0.1 * Asterisk 1.6.2.9 I'm trying to configure a new box to replace a legacy system (same hardware; some old version of Asterisk with Zaptel; works lovely but hopelessly out-of-date) and not having much joy. Specifically, I couldn't get it to see a D-channel on

i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

Hi everybody, we use Asterisk to route all calls to a inbound phone number to a specific outbund mobile phone number, depending on time and date. I'd like to send a notification email to a specific email address, each time we receive a call. For this I used the tip of "dicko" here [1]. I'm a Asterisk newbie. Unfortunately it doesn't work. The System() command is not

Hi all, I am having trouble setting the signalling method for the B410P using DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or 'bri_net' - but it doesn't mind having 'pri_cpe' etc. ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling method 'bri_net' Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 Asterisk - 1.4.23.1 Libpri - 1.4.9

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Hi I am trying to deploy freeswitch with Digium TE121 card for my office setup, but it is continuously showing Signaling is up and channels are down except D channel. Our Architecture is like We have freeswitch installed with libpri1.4 and Dahdi. I am from India and here we are having E1 trunk. Dahdi Configuration is cat system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 7

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I have tried a few different things to get asterisk to place the call in an answered state and send the DTMF 1 with the Dial macro. I found Malcom Davenports wiki page regarding Google calling which has been very helpful in troubleshooting the issue.