similar to: Playback(something,noanswer) on Zap?

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

Hi, I would to send a message to an incoming call with no answer. My Asterisk server receive the incoming call by a BRI/SIP gateway (Could be a PRI, for instance). I do the command playback with option noanswer, Asterisk send 183 followed by RTP and finish with 603. But the BRI gateway do not allow to pass the RTP without a 200 OK. The question is: are there a SIP command to indicate the gateway

Hi, according to the wiki the value NOANSWER for the channel variable DIALSTATUS means: No answer. The dial command reached its number, the number rang for too long, then the dial timed out. In out dialplan we grap all these events with exten => s-NOANSWER,1,Playback(sometext) exten => s-NOANSWER,2,WAIT(1) exten => s-NOANSWER,3,Hangup() The dial commands for internal and external

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Hello, I'm doing lot of international calls via Sixtel and VoipJet. And there are some calls which do not go through - Asterisk immediatelly returns with NOANSWER. And it is not because the dialed party does not pickup the phone, it is because the call does not go through the provider. I've written a dial macro which route the call via second provider if the first returns

Hello List, I am an Asterisk newbie, and I got a question about Asterisk Background command's option "noanswer": What is required from the user agent, such as a SIP phone, to be able to hear the playback without Answer()? I'm asking this because when I used X-Lite, I could hear the the audio file but when I used a hardware phone (an ATA in fact) I couldn't hear it. The

Hi Guys.. I am making an anoucement machine that is not allowed to "answer" the call due to a billing issue. I found that Playback with "noanwser" is usefull in this case. $AGI->exec('Playback',"$message","noanswer")} But when i request some values to the user with get_data, i think there is an answer anywere. Is there a way to get_data

Gang, I'm wondering if anyone has run into this problem and found a solution. When I use the manager interface to generate a call, I don't get very much information in my CDR records when the dial status is BUSY, FAILED, NOANSWER, etc. I am putting the dialed number into the CDR Userfield in my dialplan, but the field doesn't populate the CDR record unless the Originate action is

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4...

Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working?

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

hi,all i find in asterisk 1.6.2.1, before play a sound file use playback or background, it will answer the channel first. but i want to answer the channel when dial someone and pick up the phone.not play a file. i know there are some params such as 'noanswer' for playback or 'n' for background can not answer before play a file. but it is not always take effect on my tests.as it

All, Alright, I've looked around the internet, the voip-info.org wiki, and browsed the contents of this mailing list. While I've found a couple of scenarios that are close to this one, I haven't found one that uses my particular card (T100P). Without further delay -- I have successfully configured internal SIP services between a Snom 200 and a Windows X-Lite client and have

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

Hallo! Finally we have E1 PRI connected to our Asterisk box. Now I have one question. My internal extensions (_XXX) are SIP phones connected to Asterisk. Our telco routes some public numbers (_71602XX and others) to our Asterisk via E1. Some internal extensions can be reached from outside using public numbers (e.g. 7160234 -> 200), and some others cannot. Everyone can call outside

Hello, I have an ISDN PRI E1. I want to send an audio before answering, I am using noanswer option in playback app but all the audio is muted before the answer. I would like to play this audio. Regards, ia

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

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when