similar to: Which SIP method/option to display a short text message ?

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

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

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

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 >

Hello, My Asterisk 16.2 instance (Debian Buster package) has: same = n,Verbose(0,CHANNEL is ${CHANNEL}) same = n,Verbose(0,CHANNEL(accountcode) is ${CHANNEL(accountcode)}) same = n,Verbose(0,CHANNEL(contact) is ${CHANNEL(contact)}) same = n,Verbose(0,CHANNEL(endpoint) is ${CHANNEL(endpoint)}) and prints: CHANNEL is PJSIP/9150-00000016 CHANNEL(accountcode) is GENERAL CHANNEL(contact) is

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

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

Hi, Is is possible to "translate" non-english text into ASCII text so that SIP phones would correctly display non-ASCII characters received from SendText() ? I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines "text/plain" Content-type but googling, I can't find a source describing what text/plain can or cannot be. Regards -------------- next part

Hi all, I've got 4 actors on my stage: Alice calling from outside Bob transferring incoming calls to Charlie Charlie who has a mobile phone My PBX which is connected to my ISDN line. I want Charlie to see Alice's Callerid after Bob has transferred the call as if Charlie is receiving the call from Alice, transparently. Tried to set the callerid but Charlie sees my telco line number,

On 30/01/2020 15:02, Christian, Mark wrote: > On Thu, 2020-01-30 at 12:27 +0000, Brian Candler wrote: >> As a concrete example: I want Alice to be able to login as "alice" >> and >> "www" to machines in group "webserver" (only). Also, I want Bob to >> be >> able to login as "bob" and "www" to machines in group

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

Hello Jeremy, Friday, December 19, 2014, 3:48:51 PM, you wrote: > On Fri, Dec 19, 2014 at 12:47:51PM -0500, Rufe Glick wrote: >> Hello, >> After researching the subject on the internet I concluded that Samba should take into account Unix file acls. During my tests I found the opposite. Only Unix file mode bits are respected, and file acls are ignored. If my initial assumption is