similar to: Orginate not working well with PSTN lines

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -------------------------- [number-to-voice] exten => 8765,1,playback(num:344345,say) exten => 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk.

hi folks, i hope that i will get some help about this issue,so my configuration is : X100P card ,with FXO port ,the problem is that when i send a call using Originate,every thing goes well but what i realy need is to know how can i detect the status of this channel till the called person hung up or pick up the phone or may be busy .so how can i know that those dahdi show status and the like

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

In case it wasn't obvious in the DAHDI release announcement. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140922/0bb998aa/attachment.html>

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

Using 1.6.2.1 with a TDM400, attached to internal analog phones and PSTN. When I dial out to PSTN, I cannot send tones, like push "1" for something stupid. The call itself works, but the DTMF tones fail. -- Starting simple switch on 'DAHDI/1-1' -- Executing [6258013 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [6258013 at

Hello list, is it necessary to properly answer() an incoming call ? I don't want to answer a call because the caller has to pay even if the attached SIP-phones do not answer the phone call. Because I answer() the incoming call, the caller has to pay for 60 seconds of 'ringtone'. On the other hand, sometimes an incoming call is send to a macro where the caller is given the

Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :-

Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it expects a "180 Ringing") and we have no ringback on the pstn side. Putting a

Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI. Is there a way to get the full channel name within AMI? I'm using asterisk 11.7.0 Thanks, -Justin -------------- next part -------------- An HTML attachment was

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.

I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax machine. Both are connected to a DAHDI board. I'd like to route incoming PSTN fax calls to the extension of the fax machine and process non-fax calls through different dialplan.logic. What's the best way to go about doing this? I've looked into Fax for Asterisk, bit I'm not sure that I want it or NVFax

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Hi There, We have single PRI with multiple DID numbers and its working fine in receiving call. And if you make outbound call it will send main-line CallerID (company name). Now we want individual caller id for per extensions on outbound calls. like if i call someone he will get my extension as callerid ( 617-838-XXXX) XXXX is my sip extension something like this so next time i direct get call

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

Hi everybody, I have a 1.6.0.5 asterisk system and I?m having many problems with my pstn cards. This system has a Digium B410P with 4 BRIs configured, and an OpenVox A400E. The issue is that it?s impossible to make the system recognize the A400E, despite the ?lspci? output is: 0a:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 11:00.0 ISDN controller: Digium,

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is