similar to: Advanced Codec Negotiation: Need info and uses cases

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 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

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

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

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

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

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

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

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,

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

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

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,

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

On Thu, Jan 30, 2020 at 7:11 AM Christian, Mark <mark.christian at intel.com> 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

Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf file to determine the codec to use for a call I have 2 endpoints: [Alice] disallow:all allow:ulaw,alaw,g729 [Bob] disallow:all allow:ulaw,alaw,g729 Alice calls into Asterisk on ext 100 and then we dial Bob I want to wait until Bod side codec is chosen to answer Alice and have each channel use the codec chose on 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, 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 correct and Samba do respect Unix file acls, then I am doing something wrong. Please see the setup below and point to what I am doing wrong.

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

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

Hello, I'm having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following example (sorry for its length): INVITE sips:bob at biloxi.example.com SIP/2.0 Via: SIPS/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bK74bf9 Max-Forwards: 70 To: Bob <sips:bob at biloxi.example.com> From: Alice <sips:alice at