similar to: How to find RTP address of ongoing call?

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

https://bugs.freedesktop.org/show_bug.cgi?id=82152 Priority: medium Bug ID: 82152 Assignee: nouveau at lists.freedesktop.org Summary: any OpenGL application crashes X, locks up machine with nouveau and PRIME Severity: normal Classification: Unclassified OS: Linux (All) Reporter: celticmadman at

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When a call is starting, Asterisk starts sending and receiving RTP packets. Each packet has a sequence number. 1. Instead of logging everything as rtp set debug is currently doing, is there a way to only log: - the sequence number of the first received packet, - any missing or misplaced sequence number. 2. Is there a way to log RTP debug information in a specific file or send the same

If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm

My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose

Does anyone know how to change the source IP address/Source Interface of RTP packets? Changing the SIP source IP address in sip.conf has no apparent impact on RTP. RTP traffic still uses the address assigned to the outbound interface.

Is there anyway to change the RTP binding address? I've changed the SIP binding address successfully but the setting doesn't seem to effect the RTP binding. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/85ed8291/attachment.htm

Hello Everyone, I have a new problem where when placing the call, asterisk will automatically go into music on hold until the call is connected (ie, no ringing). It was kind of confusing because sometimes `SESSION PROGRESS` takes longer than others, during this time we are in MOH. The call does eventually connect and the MOH stops. When debugging I saw the following debug message: [Sep 10

Hello, I have asterisk 1.6.1.1 box behind NAT. On the same local network I've SIP proxy server too. The problem appears with RTP.My provider's RTP IP addresses are public. When asterisk sends SIP invite to SIP proxy, it defines local RTP IP, but not externIP. Maybe somebody knows how to solve this problem? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part

Hello, We have a system that interoperates with an external service, so that the basic call flow is: PSTN origination -> Asterisk A -> External service -> Asterisk B Initially the SDP from the external service tells the two Asterisks to send RTP directly to each other. Part way through the call the external service sends re-INVITEs both Asterisks to change the address for audio to

The question: Is it possible to change the RTP binding address? If no, does anyone have any ideas how to work around the problem? The network: 192.168.11.1 | < Asterisk > <Freeswan> < Iptables > | \

On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from the documentation the values "no" and

Hi Joshua, Could you confirm if the 5 second period for learning a new audio stream is a minimum or a maximum? The unusual call flow in question results in Asterisk learning a new audio stream when we don't want it to, and having a minimum of say 2 seconds of audio would help avoid this. Thank you! On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp <jcolp at sangoma.com> wrote: > On

Hi there, I'm having problems with the Grandstream Budgetone 101 on hangup - "show channels"/"show channels concise" output is still showing the call's channels as active. The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this, or better, does anyone have a fix? :) Thanks, David. --

Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?) -------- Weitergeleitete Nachricht -------- Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Thu, 11 Dec 2014 15:34:39 +0100 Von: Markus <universe at truemetal.org> An: universe at truemetal.org Geschenke Moritz: dunkle Schokolade. Geschenke Anna: normale Schokolade. -------- Weitergeleitete

It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum

Hi, We're trying out Dovecot to see if it's a good replacement for UW-imapd. It seems to be very slow in opening an mbox file, even after it's been indexed. (I mean way slower than UW) Here's some info on the system: Dovecot-1.0-test61 SuSE 8.1, Linux kernel 2.4 Using NFS to access mail. I've tried turning off mmap, using dotlocking, using fcntl locking (lockd, etc. are

Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like