search for: function_channel

>> There are SOME protocol level things accessible using CHANNEL[1] but that's it. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL I am trying to use the CHANNEL function listed above from the AMI. Since it is not an AMI “action”, but rather a dialplan “function”, I’m trying to figure out how to call this from the AMI. Using a telnet session to the AMI I’ve tried variations of: action: command actionid: id123 co...

...Variable: CHANNEL(pjsip,call-id) Response: Success ActionID: act1 Variable: CHANNEL(pjsip,call-id) Value: 4decf884e3ae74595906283a74f7154e at 0.0.0.0 As well, can I request many pieces of data at once? The syntax on this page (https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL) seems to suggest you pass a single parameter, “item”, yet passing just call-id did not work. I had to pass “pjsip,call-id”. Is the first parameter a category and the second the detailed item? What if I want to retrieve multiple items (or all “pjsip” items)? -------------- next part -----------...

...00] func_channel.c: Unknown or unavailable item requested: 'reversecharge' One of the lines triggering the warning at extensions.conf is: exten => s,n,ExecIf($['${CHANNEL(reversecharge)}' = '1']?Set(COLLECT_CALL=1)) https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL says that nothing has changed from 1.8 to 11. What am I doing wrong? Why 'reversecharge' is an unavailable item now? Some missing library/package? Regards Ethy

Hi all I'd like to log the client IP addr and port used for SIP and RTP *during* in a call. The IPs must be the real source IPs (internet accessible). How are these parameters available from dialplan? For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. I need the external IP:port Regards Ethy

...e bridged call) INVITE ? Or will PJSIP_HEADERS() in fact > return the header from either inbound out outbound legs? > The answer is, you can't. There are SOME protocol level things accessible using CHANNEL[1] but that's it. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230626/84bd7919/attachment.html>

...f the bridged call) INVITE ? Or will PJSIP_HEADERS() in fact return the header from either inbound out outbound legs? The answer is, you can't. There are SOME protocol level things accessible using CHANNEL[1] but that's it. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com <http://www.sangoma.com> and www.asterisk.org <http://www.asterisk.org> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pi...

...be in events, or explicitly retrieved[5]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL [4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel [5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www....

...read,Call-ID) How are you attempting to use the channel function [2]? For instance the following returned the call-id for me as well: CHANNEL(pjsip,call-id) [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_HEADER [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CHANNEL -- Kevin Harwell Senior Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200122/11ab3bef/attach...

Hi. I'd like to get the call-id from a call (the *C-0000061c* part) to set in on a field on cdr's, like [Oct 8 16:09:37] VERBOSE[25701][*C-0000061c*] app_queue.c: SIP/12234-00006b18 is busy [Oct 8 16:09:37] VERBOSE[25701][*C-0000061c*] app_queue.c: Nobody picked up in 0 ms [Oct 8 16:09:38] VERBOSE[25843][*C-00000622*] netsock2.c: Using SIP RTP CoS mark 5 ... same =>

Based on postings it should be possible to get the SIP Call-ID header value from the ARI. At what point is this value available ? As well, how do I retrieve that value - something like GET /channels/{channelId}/pjsip_header?key=Call-Id But that doesn't work. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Sun, Jul 2, 2023 at 4:18 PM TTT <lists at telium.io> wrote: > >> There are SOME protocol level things accessible using CHANNEL[1] but > that's it. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL > > > > > > I am trying to use the CHANNEL function listed above from the AMI. Since > it is not an AMI “action”, but rather a dialplan “function”, I’m trying to > figure out how to call this from the AMI. Using a telnet session to the > AMI I’ve tried variations of: &g...

On Sun, Jul 2, 2023 at 4:39 PM TTT <lists at telium.io> wrote: > >> You use the AMI action Getvar[1] which allows channel variables and > dialplan functions. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar > > > > > I actually tried that, and although I get “success” I never get useful > data. For

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

Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within

...events, or explicitly retrieved[5]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL [4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel [5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us ou...

It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVITE ? Or will PJSIP_HEADERS() in fact return the

I am not sure if its just me, but i am able to get only local channel variables containing RTCP QOS values. The Version is 1.8.14. I want to store values of bridged channel in CDR. Phone is Cisco 7941 SIP and with sip show channelstats i see all the relevant information (jitter,packet loss) i want to get. It even calculates packet loss in %. But i am not able to store it to CDR. Asterisk 1.4