similar to: Question about media before connect

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

Dear friends I need to loopback the audio on my channel. Did anybody on the development team thought about a function or app that would do that? If it is not clear, I mean that whatever audio I get, I send back. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130520/e50219b2/attachment.htm>

I am quite surprised about the degree of surprise in the group. A few days ago, somebody called a school and issued a threat, through my network. The call came from China, but of course it was US caller. The DA wants to know where call came from. The caller ID is "Restricted" and the chinese carrier is playing games. If I had a way to store the media IP, I would be able to pinpoint the

Hi Philip, Unfortunately, * speaks MGCP only as the Call Agent, rather than as the Media Gateway. MGCP is a master/slave protocol, and it would take some effort to make * work as the slave. I have the same problem: Free Telecom here in Paris includes MGCP service with their DSL. You can call any fixed phone in France at no charge! Rates to mobiles and international are quite aggressive, too.

me asking Jack Moffit: > > (i.e. are you indicating there is already a Real plugin?) Jacks' response: > Yes. Has been for almost a year. It's not finished, but it works. > Let me know what you find. I'd love for us to work together and not > duplicate efforts. Oh, I did not know that ;) If the Real Plugin had been part of the Vorbis SDK, I wouldn't even have

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

Hi there, I am seeing some very interesting thing with the latest Zaptel 1.2.X, hope may be someone can shed some light on this. Normally, to dial via your Zaptel T1 card, you would do something like: ;Dial to PSTN exten => _9.,1,Dial(Zap/g1d/{EXTEN:1}) by not adding any option after the extension e.g. no "r" and no "m", Asterisk will pass thru the session media from the

I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling

Hi all - Can anyone tell me why stylesheet_link_tag defaults to :media => ''screen'' instead of ''all'' (or just leaving it out alltogether). Seems odd since I would think by default I''d want my page printed out the same way it looked on the screen... Anyone know why that default was chosen? -philip

Is there a guide on how to use PJSIP and never have the media travel inside Asterisk? No matter what I do, I cannot make this work. Philip Orleans -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200213/c80f007e/attachment.html>

I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type "sip set debug on" Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip

Another issue of Asterisk Summer News, delivered right to your mailbox! Back here in Sweden, it's finally summer weather. Sunshine and some heat. It's good for our ice bears and the snow houses to get some sunshine :-) Asterisk development and IRC chat has gone into a lazy summer mode, but the mailing list is still cooking. It's impossible to keep up with it, for both gurus and

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

I want to compile Asterisk always for the native architecture of the machine, and I find that it is never available. It says Depends on: native_arch(E) Can use: N/A Conflicts with: N/A Support Level: core This is Fedora 20 gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1) many thanks Philip

Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150117/f148cad7/attachment.html>

I get tons of these messages chan_sip.c:10088 process_sdp: Declining non-primary audio stream: audio 30660 RTP/AVP 4 101 13 What does it mean and does it show a problem like one-way audio? Thanks for your help.

To be removed from this mailing list, simply hit reply and type REMOVE in the subject line. Thank you, T3 Communications. ********************************************************* Blank CD-Recordable media special. 650Mb, 74 min. CD-R blanks in jewel case. ONLY $2.79 Each with minimum order of 50 pieces. Shipping and handling not included. Major brands in stock. For more information on

In channel PJSIP I use this format Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss) what would be the equivalent of this format in old SIP? I tried Dial(SIP/peer/${EXTEN}@ip.add.re.ss) but it does not work. I just cannot embed the IP address in the peer's definition, but I need to use some other configuration features that are unique to each peer.

Is it possible to make cmd_dial() bridge the audio going out to the network back to the calling party as soon as dial() starts? Put another way, is it possible to have the caller hear the outside dialtone and subsequent DTMF digits? I notice that there is an option 'r' to dial(), thus: r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP