similar to: AGI + Dial + stream file ?

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When I get a SIP INVITE as follows: INVITE sip:s at 10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212 at sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX at IP:5060> Contact: <sip:1053212 at IP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71 at sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL,

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter 80.236.215.61 109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:xxxx at 80.236.215.61:64946;ob

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/a8d923ae/attachment.htm

Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-0000004d == Spawn extension (dialin, 065939191, 2) exited non-zero on

Hi, Where is the mount option 'limit-rate' in the current version? I checked in cfgfile.c and in the documentation, no mention. Yet this option did exist at one time: http://lists.xiph.org/pipermail/icecast/2010-October/011703.html http://lists.xiph.org/pipermail/icecast/2009-January/011391.html I try to limit the bitrate of a mount-point, is there another solution? Do you know why this

Hello, Thank you for your response. It is on the kh version.. https://github.com/karlheyes/icecast-kh Le sam. 3 nov. 2018 à 21:47, Thomas B. Rücker <thomas at ruecker.fi> a écrit : > Hi, > > On 11/03/2018 07:33 PM, Mickael MONSIEUR wrote: > > Hi, > > Where is the mount option 'limit-rate' in the current version? > > I checked in cfgfile.c and in the

What is the interest to supply binary of Asterisk, under debian for example, while to use MeetMe it is necessary to COMPILE Asterisk ??? :-)) Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100611/e4d749f6/attachment.htm

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, The MeetMe application refuses MusicOnHold personalized and skip always in the default! Have you any idea how to fix this? -- Executing [028883899 at default:1] Set("SIP/109.10.214.1-00000002", "CHANNEL(language)=fr") in new stack -- Executing [028883899 at default:2] Answer("SIP/109.10.214.1-00000002", "") in new stack -- Executing

Hello, Is it possible to connect two *callers* without going through a conference (meetme) ? Example: 06:50pm - User 1 call extension 600 and musiconhold / parked call .. 06:51pm - User 2 call extension 600 and connect to User 1. Thank you in advance, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <-> SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I am having a bit of problem getting MusicOnhold to play. I am running Asterisk 1.4 with MPG123 0.59 installed. And here's what i see in the debugging window of asterisk: -- Started music on hold, class 'default', on channel 'SIP/x123-082043d0' -- Stopped music on hold on SIP/x123-082043d0 Any idea why it is not playing the file at all? thanks

I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Thanks! jlc

I have a TE210P digium card that has 2 E1/T1 ports. the code in my extensions.conf file for span 1 is : [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 ; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This

Hey, all. I was just wondering if there were any tools/utilities/what-have-you out there that would allow a user to click on a contact in Outlook, and have their phone dial it? (Or, I guess, have Asterisk dial both their phone and the destination number, and put the two into a conference.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is