similar to: DMTF payload bug in 13.14.1 with pjsip and direct_media?

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

On Mon, Jan 27, 2020 at 4:31 PM Eli Friedman <efriedma at quicinc.com> wrote: > I assume by “drop support”, you mean reject it in the bitcode reader/IR > parser? We can’t reasonably support a complex feature like inalloca if > nobody is testing it. If we can’t reasonably upgrade it, and we don’t think > there are any users other than clang targeting 32-bit Windows, probably

On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com> wrote: > Hey guys, > > have issues with reinvite, no matter what endpoint is calling asterisk > always tries switch simple_bridge to native_rtp > > Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge > technology to native_rtp > > in endpoints table ?direct_media? sets to ?no? on

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote: > NAT endpoint calling local endpount - switching to native_rtp then no audio, > both of them have direct_media=no, Verbose log: > > -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in > new stack > -- Launched AGI Script /pbx/agi.php > -- AGI

Sorry for the delay. Arthur Eubanks has started working on the design here: https://reviews.llvm.org/D74651 I felt I should follow up here about that. On Mon, Jan 27, 2020 at 6:47 PM Eli Friedman <efriedma at quicinc.com> wrote: > It doesn’t seem like multiple call sites should be a problem if they’re > sufficiently similar? If the argument layout for each callsite is the > same,

Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table ?direct_media? sets to ?no? on all endpoints but it doesn?t help. if native_rtp not work for some reason I have oneway audio. how can I fix this?

On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com> wrote: > Well, it breaks audio for all NAT endpoints, how can I fix this? > Local (packet to packet) bridging should not do that. Remote (direct media) can do that. Can you confirm - by looking at a verbose level 4 log - how Asterisk is bridging the two channels? -- Matthew Jordan Digium, Inc. | Director of Technology

Hi, How to accept DMTF tone during ringing mode? Its possible. Regards -Hadi.Salem

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

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries

I'm having problems with a new asterisk PBX install. the phones/ATAs are all linksys/cisco. They all worked before with a commercial softswitch. Most of the linksys devices offer auto, inband, INFO and AVT. I'm looking for suggestions. Thanks in advance -- One day at a time, one second if that's what it takes

Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some

Hello, Does anyone know whether chan_sip in Asterisk supports DTMF in clock rates other than 8000? I looked for telephone-event/16000 in the changelog and in Jira but no luck. Any help would be appreciated. -- Best regards, Vlasis Chatzistavrou.

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

Hello all, A few years ago, I added the inalloca feature to LLVM IR so that Clang could be C++ ABI compatible with MSVC on 32-bit x86. The feature works, but there is room for improvement. I recently took the time to write up a design using token values that will hopefully be better named and easier to work with and around. For the technical details of the proposal, I've written up the RFC

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello-- Why do I target chan_sip for so much effort? Because, it seems to me, chan_sip is probably the most used channel driver in the asterisk community!! (and, of course, the zap/dahdi driver, is also pretty popular) I haven't had time to follow up on chan_sip, and I probably won't for several months. But, if I had time, here is what I'd do: There are two ways to speed up

hello, i've an asterisk box which is connected to an E1/PRI via a TE110P card. incoming calls from mobile phones where the number is transfered as a whole block work fine, but when dialing from an analog or ISDN line to the asterisk box there is a timeout of about 3-5 seconds. originally my incoming context looked like: exten => _X.,1,Dial(SIP/${EXTEN}@domain.tld) so i assumed that the

Hello, The other day, a 13.14.1 server suddenly stopped working correctly. First, it printed: Oct 23 21:53:40 FOOBAR asterisk[2377]: WARNING[27942]: db.c:332 in ast_db_put: Couldn't execute statment: SQL logic error or missing database This occurred while this server received a lot incoming REGISTER such as: Oct 23 21:53:40 FOOBAR asterisk[2377]: [Oct 23 21:53:40]