similar to: PJSIP redirect_method=uri_core and header modifications

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? 2 channels below are somehow associated, but how can I be 100% sure they are related in order to implement a redirect of the incoming call to another phone ("attended" call pickup respecting call/pickupgroups). Uniqueid seems to be a

The relevant part of my setup is something like: SIP phones -> local server -> remote server -> SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. Do I need to set up two separate IAX2 connections, one "privileged" and the other not, or can I

Assuming I've got a setup where calls entering Asterisk on SIP leave on IAX2 ( and the reverse), i.e. a SIP user might dial '1234' where we then have extern => 1234,1,Dial(IAX2/somewhereelse) Now, we don't have any G.729 functionality on this server, so what happens if the SIP user calls with G.729 only available? Assuming the remote IAX2 server does have G.729 can it be

Somehow I can't get 1.6.2.13 to compile with DEBUG_CHANNEL_LOCKS. Downloaded latest tgz and extracted $ ./configure $ make menuselect (select the needed options from compiler flags) $ grep DEBUG_CHANNEL_LOCKS menuselect.makeopts MENUSELECT_CFLAGS=DONT_OPTIMIZE LOADABLE_MODULES DEBUG_CHANNEL_LOCKS MALLOC_DEBUG $ make && make install $ asterisk && asterisk -rx "core show

Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: root at Sangoma-Testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? Thanks! -------------- next

Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the "REAL" machine? Thanks Danny Nicholas

I was wondering why *foo at default should match '_*[0-9a-zA-Z].*0.' in 1.6.13. Who is making the parse error, * or me? CLI> dialplan show *foo at default '_*[0-9a-zA-Z].*0.' => 1. NoOp(${EXTEN}) [pbx_config] 2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config] 3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]

I'm trying to figure out how to commit some code for review. Following: https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage Created a ssh alias. Cloned using: "git clone ssh://asterisk/asterisk" Set name and email. Installed the gerrit commit hook: "git review -s" Try to change to asterisk 13 for creating a patch: "git checkout 13" This fails with: error:

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

Hello asterisk users, I have a strange behaviour with asterisk and error code forwarding in asterisk 11. Please find below my setup: Phone -> ASTERISK -> SIP TRUNK PROVIDER A phone start a call, asterisk start a leg to my SIP trunk provider. I have a simple dialplan to handle it: [gotoexternal] exten => _X.,1,Dial(SIP/${EXTEN}@provider) When my SIP provider return to asterisk a 404

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP

> On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote: >> I want to capture all SIP messages. >> >> I have about 30 hosts in about 6 colos. >> >> My first thought was dumpcap, but the output file name format bugs me. >> >> What do you use for long term SIP capture? On Wed, 31 May 2017, Daniel Tryba wrote: > What bugs you about the output

On Wed, 31 May 2017, Daniel Tryba wrote: > On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote: >>> What bugs you about the output format? >> >> It's been a while, but as I recollect, it included the date/timestamp in the >> file name of the 'ring buffer' which meant that each time the host was >> rebooted, dumpcap didn't know the

I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register.

On 05/17/2018 11:38 AM, Frank Vanoni wrote: > On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote: > >> 3. How do I set up the server to block these ? >> >> 4. Can I stop the retransmitting of the 401 Unauthorized packets ? > > I'm happy with Fail2Ban protecting my Asterisk 13. Here is my > configuration: > > in /etc/asterisk/logger.conf: > >

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

OK - I'll have to rethink how to solve this problem. Maybe I made some assumptions...here's what I'm trying to accomplish: I've been given a legacy PBX with SIP capabilities. I need to have SIP phones connect to Asterisk (for other features, part of the next step) which passes the calls through to the legacy PBX. And conversely, calls to that extension number on the legacy PBX

Hey Guys, I'm after a way to monitor our sip trunks (peers) and send an email if they go down. I know I could use 'asterisk -rx "sip show peers"' in a shell script but that seems messy, especially since I'd like to monitor it fairly closely (so I'd like to run it every 20 or 30 seconds or so). Is there a better way to do it?

Hi, I tried to perform call forward in asterisk by writing the following in the dial plan.The data base is getting updated with the caller ID number how ever the call is not getting forwarded. [apps] exten => _*21*XX,1,Set(DB(CFIM/${CALLERID(number)})=${EXTEN:4}) exten => _*21*XX,2,Hangup exten => #21#,1,DBDel(CFIM/${CALLERID(num)})=${EXTEN:4} exten => #21#,2,Hangup Regards, Aparna