similar to: Example of ${CHANNEL(contact)} output ?

Greetings All, We've been working hard on new codec negotiation stuff for Asterisk 18 and we've got some stuff to run by you. It's a lot so please read carefully. To give you some idea of just how difficult a job this is, a simple call from Alice to Bob currently causes 8 attempts to reconcile codecs between them in app_dial, chan_pjsip, res_pjsip_session and res_pjsip_sdp_rtp. If

El Tue, 9 Jun 2020 09:46:32 -0600 George Joseph <gjoseph at digium.com> escribió: Hi George > > > > > > If transcoding is enabled Would it be possible to do the same but handle a > > 488 > > back from Bob and failover to another INVITE with Bob's allow list to > > handle > > transcoding? That way we would always try no-transcoding before

I have wrote a small Linux Shell command for implementing Interlock Protocol which is known as a cryptographic protocol that resistant to man-in-the-middle attack. Here is the steps of interlock protocol: *(1)* Alice send her public key to Bob *(2)* Bob send his public key to Alice. *(3)* Alice encrypts her message using Bob's public key. Then she sends half of that encrypted message to

>> So which option is preferred? >> >> A) Have a softphone aor/auth_user/password for a particular human, and >> expect them to configure it on multiple devices. Do not worry that 1) >> multiple are registered at once (because that's normal in SIP) and 2) >> asterisk has no idea which is which (because the intent is to place a >> call to

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

Hi, Using SIP methods and options, is there any way for a callee to send the caller a short text message when the call is establishing ? Scenario is : Alice and Bob's SIP phones are registered to an Asterisk server. Alice calls Bob : an INVITE message is sent to Bob's phone Bob is replying : a 200 OK message is sent back to Alice with a short text included ("Welcome to BoB

On Dec 19, 2014 9:05 PM, "Rufe Glick" <rufe.glick at gmail.com> wrote: > > Hello Jeremy, > > Friday, December 19, 2014, 7:00:06 PM, you wrote: > > > On Fri, Dec 19, 2014 at 06:31:33PM -0500, Rufe Glick wrote: > >> Hello Jeremy, > > >> Friday, December 19, 2014, 4:55:21 PM, you wrote: > > >> > On Fri, Dec 19, 2014 at

Dear List It's probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multiple AOR, so you need to expand them with ${PJSIP_DIAL_CONTACTS()} to be able to Dial() all of them simultaneously. But there are also

Hello Jeremy, Friday, December 19, 2014, 7:00:06 PM, you wrote: > On Fri, Dec 19, 2014 at 06:31:33PM -0500, Rufe Glick wrote: >> Hello Jeremy, >> Friday, December 19, 2014, 4:55:21 PM, you wrote: >> > On Fri, Dec 19, 2014 at 03:58:58PM -0500, Rufe Glick wrote: >> >> Hello Jeremy, >> >> > Do alice and bob have the same user ids on client

Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When Bob forwards to Cory a call coming from Alice, would expect Diversion/History-Info header to

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

It would appear that you have the Asterisk server on a public IP address, your two endpoints are behind a NAT, and you have rewrite_contact enabled in pjsip.conf. In which case, what you are seeing is correct. In order to be able to send a call to an extension where it is behind NAT, Asterisk must update the contact to have the current IP and port that the phone registered via (i.e. the WAN IP

Hello Jeremy, Friday, December 19, 2014, 4:55:21 PM, you wrote: > On Fri, Dec 19, 2014 at 03:58:58PM -0500, Rufe Glick wrote: >> Hello Jeremy, >> > Do alice and bob have the same user ids on client >> > and server ? >> Yes, the uids and gids are identical on both server and client machines. > Then it should work. Set debug level 10 on the smbd > and look

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

On 1/30/20 5:48 PM, Christian, Mark wrote: > On Thu, 2020-01-30 at 16:37 +0000, Brian Candler wrote: >> I was hoping to avoid the dependency on configuration management by >> carrying the authorization in the certs themselves - if that is in >> the spirit of the SSH cert mechanism. > > Sign alice and bob's ssh cert with principal's alice,www and bob,www >

Antonio G?mez Soto wrote: > > So basically, the 'contact' in the AOR is just an ip address (or > 'dynamic', in which case it accepts > incoming registrations). A contact is a SIP term, it's a way of getting to something. (IP address+port) > So what happens if one endpoint has multiple AOR's which are registered > from different ip addresses. > And

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24 network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is

On Fri, Dec 19, 2014 at 06:31:33PM -0500, Rufe Glick wrote: > Hello Jeremy, > > Friday, December 19, 2014, 4:55:21 PM, you wrote: > > > On Fri, Dec 19, 2014 at 03:58:58PM -0500, Rufe Glick wrote: > >> Hello Jeremy, > > >> > Do alice and bob have the same user ids on client > >> > and server ? > > >> Yes, the uids and gids are