similar to: Indications missing on Cisco FXO -> ATA-186 (SIP)

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

Hi, try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. A known problem ??? Thanks in advance Michael i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus) and oh323-0.6.0 Here are my config's ############## # modem.conf # ##############

Hi I use dial with music on hold command exten => _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem if the called party line is closed or number is incorrect or have a voice mail (Early media 183) user will not hear the message from operator notifying that line is out of service , temporary unavailable ., what to do to solve this problem In other words how to stop MOH since

Hi, I have a queue that plays music when a call comes in. To be able to do that I need to Answer() the call first. After a timeout in this scenario the call should be transfered to an extension using a GoTo statement to the extensions context. The problem is that as soon as asterisk Answers the call it can not play a ringtone (or other tones) back to the original caller when executing a Dial

Hi, I've looked around the archives and have spent a while on voip-info.org but not found an answer so forgive me if this is in a FAQ somewhere. We've got several Asterisk servers with E1 cards (some Digium, some Sangoma). We provide non geographic numbers for customers and route calls to their existing phone numbers. Calls come in over the PSTN and into Asterisk. This works perfectly

i am creating call files, and catching successfully the ones that don't connect in a 'failed' extension. can anyone tell me how to find out the reason for the failure (ie busy, no answer). ${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't used) and channel_status doesn't seem to be any good. thanks in advance. -- - Rich Doughty

Hi, The descriptions of Progress() and Proceeding() are really vague. Progress(): ---cut---------------- [Synopsis] Indicate progress [Description] Progress(): This application will request that in-band progress information be provided to the calling channel. ---cut---------------- Proceeding(): ---cut---------------- [Synopsis] Indicate proceeding [Description] Proceeding(): This

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

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

Hi there, Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

Hi, I have the panel data of income statement of several banks. The date 9803 means 1998-Q1, 9806 means 1998-Q2, etc. I transform the date code to 1 (for 1998Q1), 2 (for 1998Q2), ...., 16 (for 2011Q4) where 1, 2, .... are placed in Col1. Now the income statement of a specific quarter is actually the accumulation from the beginning of the year. For example, the cost data of 1999Q3 is the

hello list... i have configured my asterisk in such a way that it first screens an incomming call and the caller has to enter his/her pin number so that he is connected to the system. i am writing a prepaid application to incorporate into the asterisk PBX. however, after searching the database for the user's pin number, the dial application on my dialplan does not work properly. it gives

Bugs item #5503, was opened at 2006-08-23 21:27 You can respond by visiting: http://rubyforge.org/tracker/?func=detail&atid=411&aid=5503&group_id=85 Category: win32-process Group: Code Status: Open Resolution: None Priority: 3 Submitted By: David Haney (darius42) Assigned to: Nobody (None) Summary: Process::create() checks return value of CreateProcess against 0, should be false

Hi All, I have a question about Fastagi because I can't get it to work for some reason. Everytime I execute the fastagi command, i get an error: my extensions.conf: .. exten => 1000,1,agi(agi://some_ip_address) .. output from asterisk console: -- Executing AGI("Zap/1-1", "agi://some_ip_address") in new stack -- Launched AGI Script

Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: "SPA922" <sip:5403@mysipserver.com>;tag=685bbad1fae3325do0 To:

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer or make a three way call. The Zap/x-2 channel is created and the transfer or three way proceeds, but on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk goes crazy logging the problem. Here is an example debug log. This happens only once a day or so, with 100 or so users transfering and three

Hi, I need help finding why my system is dropping calls.. I enabled debugging on my box in the hope it would lead me to the answer as to why my system is dropping calls but unfortunately nothing is jumping out at me.. I have attached the portion of the messages file for two calls that were dropped.. (numbers names and ip's have been changes to protect the innocent) Can someone give me a

Hi all, i have noticed that i can't hear a ring tone if i make a call from my TDM40B to a chan_modem_i4l endpoint. I had a look in the code from chan_modem_i4l and there is a function calling "i4l_handle_escape" that gives a AST_CONTROL_RINGING frame back. But this seems not work ...(or i4l is not signaling it ?) Til now i have used the Dail app like Dial, Zap/g1:XXXXXX|60|r so it