similar to: DTMF fun

Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, -JK JK, In-band or RFC2833 DTMF signaling? Also, unless you have SER configured with a media proxy, the actual "call" is not running through SER. It's a signaling proxy only. -- Alex Balashov Evariste Systems Web :

Sorry if this is an obvious question and perhaps my Google foo isn't right on this one: I have calls coming into an Asterisk server that may be using 2 different codecs. I am recording audio in both cases but the challenge is knowing which codec was negotiated at call setup. I need to pass the proper format to the record command as the codecs cannot be transcoded and are only supported for

Hello Masters Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Hi, Few subjects cause as many arguments as "which SIP client works best?" on IRC #asterisk, voip forums, and probably the -users mailing list. I have tried most of the SIP clients available in the last 5 years, both with Asterisk and other platforms such as OnSIP.com, IConnectHere.com, ZipDX.com and the venerable old FWD (in the days when that almost worked). Speaking of ZipDX, we

asterisk-users-request@lists.digium.com wrote: > Date: Sun, 22 Apr 2007 19:38:04 +1000 > From: Rob Hillis <rob@hillis.dyndns.org> > Subject: Re: [asterisk-users] Softphone that supports central > provisioning? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <462B2CFC.50709@hillis.dyndns.org>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List (Again) Thanks to some answers on this list and another I now have a MultiTenant system/setup working the way that I want it to, So now my next job is to find a SIP SoftPhone that I can brand to my own company images and so on. Again an OSS would be preferred, Even though X-Lite is bar far one of the best free SIP clients I have used and it

Test both the TLS enabled and fallback paths. nbd-server doesn't appear to support TLS at all, and qemu-nbd is known not to allow fallback to unencrypted, and therefore it only makes sense to test nbdkit at the moment. --- .gitignore | 4 ++++ TODO | 3 --- interop/Makefile.am | 54 +++++++++++++++++++++++++++++++++++++++++++++ interop/interop.c | 30

Hi, I know the following is mostly the issue of SER and I already posted the same content to SER User list. Just for more input, I posted it to this list. Sorry for the cross post for some people. I've set up SER for UA to UA call. I'm thinking of setting up SER to relay to Asterisk PBX to use conference call and voicemail of Asterisk. I will employ this system for client connection

Hi, In the aftermath of Digium's and Counterpath's Bria for Asterisk announcement, we're happy to chat with Todd Carothers, Counterpath Product Manager today at 1 PM EST. For more info, http://vuc.me Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc Call in starting at around 12 Noon EST: sip:200901 at login.zipdx.com Hear you there! /r

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

On Fedora 29, nbdkit-devel at 1.10.4 is too old for any of the tests/Makefile.am actions to run, but this also means we don't create any tls pki/ or keys.psk files that we then want to reuse during qemu testing, as evidenced by 'make check' failures such as: FAIL: interop-qemu-nbd-tls-certs ================================ qemu-nbd: Unable to access credentials

This neutral refactoring adds -DTLS_MODE. We can in future change the requested TLS mode, but not in this commit. It also checks that nbd_get_tls_negotiated returns true after connecting, when the requested mode was set to LIBNBD_TLS_REQUIRE. --- interop/Makefile.am | 4 ++++ interop/interop.c | 26 ++++++++++++++++++++------ 2 files changed, 24 insertions(+), 6 deletions(-) diff --git

The test interop-qemu-nbd-tls-certs frequently fails on slow (32 bit) machines in Fedora Koji. (Is crypto slow on these already overloaded machines?) As we cannot wait for a signal when qemu-nbd is ready start serving, we have to use a sleep. The current sleep is 5 seconds, which is not long enough. Making the sleep longer would work but is inconsiderate for people using faster machines.

Hi, I've noticed that asterisk seems to stop relaying DTMFs after a call has been up for a while (~10 mins). I was just wondering whether this was intentional, or a bug. In detail here's my setup SIP Gateway --> Asterisk --> E1 --> Asterisk --> SIP Gateway The LHS gateway sends RFC2833 DTMF messages to the first Asterisk which bridges them onto the E1. They then get

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Everybody, I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8 build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on LAN (without NAT). With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can have a very fine secure conversation in both directions. When I want to do the same with my iPhone,

This adds new APIs for running a local NBD server and connecting to it using systemd socket activation (instead of stdin/stdout). This includes interop tests against nbdkit and qemu-nbd which I believe are the only NBD servers supporting socket activation. (If we find others then we can add more interop tests in future.) The upstream spec for systemd socket activation is here:

The NBD protocol is adding an extension for servers to report the initial state of the image (for now, whether it is sparse and/or reads as all zeroes). Time to expose this to clients, via the new API nbd_get_init_flags(). The patch requires refreshing the nbd-protocol.h file from a corresponding contemporary nbdkit commit. Testing is possible with recent enough qemu. --- .gitignore