search for: tryba

...'ChannelStateDesc' => 'Ring', 'Context' => 'macro-dial-one', 'Extension' => 's', 'Priority' => '37', 'Seconds' => '3', 'Uniqueid' => '1286364290.8474', ) -- Daniel Tryba

...PTIMIZE LOADABLE_MODULES DEBUG_CHANNEL_LOCKS MALLOC_DEBUG $ make && make install $ asterisk && asterisk -rx "core show locks" No such command 'core show locks' (type 'core show help core' for other possible commands) Am I missing something? -- Daniel Tryba

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

...[pbx_config] 2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config] 3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config] 4. Set(CDR(accountcode)=${accountcode}) [pbx_config] 7. ResetCDR() [pbx_config] 8. ... -- Daniel Tryba

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

With chan_sip there is the variable SIP_MAX_FORWARDS to set Max-Forwards. This counter is persistant after a redirect. I can't find the equivalent for PJSIP, so I went the way of header manipulation. Only to find out that any headers added to the outbound leg are lost after a redirect (with redirect_method=uri_core (didn't try any other since in the past they didn't work for me)). Am

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:

...ve 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 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 files from the previous run should be deleted...

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 ti...

Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba <daniel at tryba.nl> To: Asterisk Users Mailing List - Non-Commercial Discussion     <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Is there any way to pass caller id to     cell phone? Message-ID: <20181015213930.2a4uulq2z6xbfjcb at bogus> Content-Type: text/p...

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

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

...rpret 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. >> >> How can I tell which codec was actually used for the call? On Fri, 11 May 2018, Daniel Tryba wrote: > AFAIK this is undetermined. The callee can send either ulaw or gsm, > unless the caller wants to narrow it down to 1 codec, see > https://tools.ietf.org/html/rfc4317#section-2.2 > > Most of the time the callee will pick the first (so in this case ulaw). > But there are m...

...tering on the legacy Sip pbx, etc. I think I'm stuck at the conceptual level. (Still a beginner in training - but having fun learning Asterisk) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel Tryba Sent: Wednesday, April 11, 2018 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Pass through registration / proxy On Tue, Apr 10, 2018 at 09:22:02PM -0400, Telium Technical Support wrote: > I need to cr...

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: > >

Can't find a way to control the dtmf mode on a per session basis with pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any hints on how to do this?

Hi list, I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI--> IAX testing. Any ideas? Google wasn't really helpful on this one. Danny Nicholas