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Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows.... - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing "a=sendonly". - Asterisk plays the caller music on hold, which I...

On 12.12.18 at 19:43 Joshua C. Colp wrote: > On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote: > > <snip> > >> >> The problem: The extension doesn't create a ringback locally, because >> it most probably expects it to >> be sent by the callee - but the callee doesn't send anything (not >> surprising, because there has been >>

...of GSM gateway (it has call waiting activated on the sim card), call is forwarded to Asterisk queue to the same extension 1000 and a pop-up appears with the second call. c) extension 1000 accepts it so put on hold first call, then try to pickup the new one. The thing is that the SIP re-invite with sendonly attribute can be seen from extension 1000 to Asterisk queue, but this SIP invite is not being forwarded to GSM gateway. So the GSM gateway keeps waiting for it and because it never appears the 1st call is dropped. Maybe you have had this issue in the past. I know that Im not an expert, but I hav...

...red SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating sendrecv on streams. * Device sends ACK with SDP indicating sendonly on streams. Now, I would like to configure an Asterisk instance to act as those SIP devices, ie to defer all SDP signaling in ACK. This is for testing purpose as I would like to reproduce in a lab an issue with those SIP devices. 1. Is it possible ? I can use any Asterisk version for implemen...

...some actions in the dialplan and should place the call on hold after some time, so that the caller only hears the on hold music from my provider (not streamed by my Asterisk). Technically speaking I want asterisk to send a re-INVITE message containing an updated SDP body with the attribute "a=sendonly" or "a=inactive" added so that the SIP server of my provider (where Asterisk is registered to as user) will recognize that the call should be placed on hold. A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Exam...

Hi all If Asterisk is bridging a call between two SIP peers and one peer puts the other on hold by means of a re-INVITE with SDP containing a=sendonly, Asterisk will play locally generated MOH instead of relaying the media streamed by the SIP peer which took the hold action. Any ideas how to change that? (This is understandable if the peer is a handset but can be a problem if it is a PBX with its own MOH source.) Richard -- Richard Brady

Hi All, When asterisk bridges a call between 2 peers and peer-A's user puts the call on hold, then peer-A sends a INVITE with recvonly in the SDP. Asterisk responds to peer-A with sendonly in the SDP and asterisk sends an INVITE to peer-B with recvonly in the SDP. Peer-B then responds with a sendonly in the SDP. I've noticed in the above scenario that peer-B contiutes to send audio to peer-A. What is the point in having audio from peer-B to peer-A as the user at peer-A has put...

Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value "a=sendrecv" is present, according to the rfc3264 the sdp value a must be mark with "sendonly". I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and directmedia Can anybody help me? Thanks a lot Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: &lt...

...archar(255) | NO | MUL | NULL | | | password | varchar(255) | NO | | NULL | | | quota | int(10) unsigned | YES | | 0 | | | enabled | tinyint(1) | YES | | 0 | | | sendonly | tinyint(1) | YES | | 0 | | | last_login | int(11) | YES | | NULL | | | last_login_ip | varchar(16) | YES | | NULL | | | last_login_date | datetime | YES | | NULL |...

...ty RTP frame) just to prove it is alive. Unfortunately Aastra said the hold behaviour on the phone is correct, as per RFC 3264, section 8.4, 4th paragraph: Typically, when a user "presses" hold, the agent will generate an offer with all streams in the SDP indicating a direction of sendonly, and it will also locally mute, so that no media is sent to the far end, and no media is played out. They can implement the "RTP keep-alive" feature only if there is some RFC describing that behaviour. Is Aastra correct? Should I configure Asterisk with rtpholdtimeout=0 to sol...

...aiting > activated on the sim card), call is forwarded to Asterisk queue to the same > extension 1000 and a pop-up appears with the second call. > c) extension 1000 accepts it so put on hold first call, then try to pickup > the new one. > > The thing is that the SIP re-invite with sendonly attribute can be seen from > extension 1000 to Asterisk queue, but this SIP invite is not being forwarded > to GSM gateway. So the GSM gateway keeps waiting for it and because it never > appears the 1st call is dropped. > > Maybe you have had this issue in the past. I know that Im...

...get extension's channel to...well, something) -- if the parking space was returned to the dialplan somehow (or if Park() didn't ignore its arguments). At the moment, I'm using the v1-0 branch and at this point it looks like our phones will be all SIP (i.e., madding chan_sip to make sendonly channels visible to the dialplan somehow isn't automatically out of the question). -- Joshua P. Dady -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3721 bytes Desc: not available Url : http://lists.digium...

...Em>): unknown user Jan 14 15:07:10 auth-worker(5346): Debug: sql(mailuser1 at mydomain.com,213.225.33.38,<MpKNj2t/V7XV4SEm>): SELECT concat('*:storage=', quota, 'M') AS quota_rule FROM accounts WHERE username = 'mailuser1' AND domain = 'mydomain.com' AND sendonly = false; Jan 14 15:07:10 auth: Debug: master userdb out: USER 1124466689??? mailuser1 at mydomain.com quota_rule=*:storage=2048M auth_token=c0af49e6da382961494c74d54add28b3a077f23c Jan 14 15:07:10 imap-login: Info: Login: user=<mailuser1 at mydomain.com>, method=PLAIN, rip=213.225.33.38,...

...14 15:07:10 auth-worker(5346): Debug: >> sql(mailuser1 at mydomain.com,213.225.33.38,<MpKNj2t/V7XV4SEm>): SELECT >> concat('*:storage=', quota, 'M') AS quota_rule FROM accounts WHERE >> username = 'mailuser1' AND domain = 'mydomain.com' AND sendonly = false; >> Jan 14 15:07:10 auth: Debug: master userdb out: USER 1124466689 >> mailuser1 at mydomain.com quota_rule=*:storage=2048M >> auth_token=c0af49e6da382961494c74d54add28b3a077f23c >> Jan 14 15:07:10 imap-login: Info: Login: >> user=<mailuser1 at mydomain...

Hi everybody, I run two redundant Dovecot servers with a shared Maildir on a GlusterFS volume and a SQL authentication backend based on a mirrored MariaDB database. Because of the splitbrain situation I would like to add two Dovecot Director as proxies. Since a few days I am trying to get the setup running. In the meantime I have achieved that the clients can successfully authenticate on the

hi, its possibe to "dont start" music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone) Asterisk acts on SDP a=sendonly i want pass trough media from SIP trunk provider Marek

...multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-16.8-cert3 Thank you for your continued support of Asterisk! ------------...

...omething like change "loopback" mode to "normal" mode?)?? Does anyone has already faced some situation like this?? I don't know what could be happening... The create connection I send is: CRCX <transaction_id> TS/trunk#0/$@stretto2k MGCP 1.0 C: 987654321 L: p:20 M: sendonly a=dialed:111222 a=called:333444 a=dialing:123456 The last 3 lines I set with "the called number, the number to which the call was delivered and the calling number" (RFC 3435). Other SDP parameters different from "a=", Stretto 2000 doesn't allow me to set. Thank...

Hello ppl, Is there anyway to control a call mid-way in terms of sending a re-INVITE with say sendonly, etc. to mute one call leg of a bridged call ?? Looked around, so far, doesnt seem to be possible. If it's not, I think it's quite an important feature (re-INVITES mid-session) for a B2BUA. cheers - Ben.

...io.net Posting Request: AUEP 3 aaln/1 at 0-13-11-82-bd-a.ssw.dominio.net MGCP 1.0 NCS 1.0 F: A to 10.30.15.254:2427 MGCP read: 200 3 OK A: a:PCMU;PCMA;G728;G729;G729E;G726-16;G726-24;G726-32;G726-40, p:10-30, b:19-100, e:on, t:1, s:off, v:L;fxr;rg;xal;x-xl;fm;lcs;sst;x-jc;x-pol;xrm, m:sendrecv;sendonly;recvonly;inactive;netwloop;netwtest;replcate;confrnce, dq-gi, sc-rtcp: 81/70;81/71;82/70;82/71;80/70;80/71, sc-rtp: 62/51;62/50;64/51;64/50;60/51;60/50 A: a:telephone-event, fmtp:"telephone-event 0-15,144,149,159" A: a:image/t38, p:10-30, b:25-64, dq-gi from 10.30.15.254:2427 Verb: &...