# No ringback for incoming calls **URL:** \ **Category:** Asterisk SIP\ **Created:** [November 23, 2023, 6:12pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771 "2023-11-23T18:12:20Z")\ **Posts on this page:** 11\ **Page:** 1
**Author:** ![Vince\_be](https://avatars.discourse-cdn.com/v4/letter/v/59ef9b/32.png) [@Vince\_be](https://community.asterisk.org/u/Vince_be)\ **Post date:** [November 23, 2023, 6:12pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/1 "2023-11-23T18:12:20Z")
Hi all, Installation of my new server is progression well (asterisk 20.5). I just noticed today an issue related to incoming calls. There is no ringback :-(. Here is the dialplan for incoming calls : ```auto [common-external](!) exten => s,1,NoOp(${CONTEXT}) same => n,NoOp(${CALLERID(num)}) same => n,Answer() same => n,Set(ARRAY(admin_name,admin_context)=${ATTENDANT_IS_ADMIN(${CALLERID(num)})}) same => n,GotoIf($["${admin_name}" = ""]?get-message-type) same => n,Playback(fr/flexia/${admin_name}) same => n,Gosub(service,s,1(${CONTEXT})) same => n,Goto(ring) same => n(get-message-type),NoOp();Set(message_type=${ATTENDANT_GET_MESSAGE_TYPE(${CONTEXT})}) same => n,GotoIf($["${message_type}" = ""]?ring) same => n,Playback(fr/flexia/${message_type}-message) same => n,GotoIf(${REGEX("^exceptional-.*$" ${message_type})}?emergency) same => n,Playback(fr/flexia/schedules-${CONTEXT}) same => n(emergency),Playback(fr/flexia/emergency-numbers-${CONTEXT}) same => n,Hangup() same => n(ring),Set(endpoints_list=${HOTDESK_GET_ENDPOINTS_LIST(${CONTEXT})}) same => n,Dial(${endpoints_list},10,b(handler^addHeaderExternal^1)) same => n,Playback(fr/flexia/wait-1) same => n,Dial(${endpoints_list},25,b(handler^addHeaderExternal^1)) same => n(wait2),Background(fr/flexia/wait-2) same => n,WaitExten(30) exten => 1,1,Dial(${endpoints_list},,b(handler^addHeaderExternal^1)) exten => 2,1,VoiceMail(${${CONTEXT}+"-mailbox@flexia"}) same => n,Hangup() exten => i,1,Goto(attente2) exten => i,n,Hangup() exten => t,n,Hangup() ``` I tried several options for the Dial application : Dial(…,r) =\> no effect Dial(…,R) =\> no effect Dial(…,rm) =\> music on hold is working (quality is very poor). Playback application in the dialplan works without any issue. I tried also to play with inband\_progress in pjsip.cong but also without success. When looking in pjsip logger, the only place where I can find a 180 Ringing is here : ```auto [2023-11-23 19:08:14] <--- Received SIP response (675 bytes) from UDP:91.179.112.248:5060 ---> [2023-11-23 19:08:14] SIP/2.0 180 Ringing [2023-11-23 19:08:14] Via: SIP/2.0/UDP XX.XX.XX.XX:12345;rport=56278;branch=z9hG4bKPj4a6de03f-2b56-4600-be2d-ab6a0f00c4fa [2023-11-23 19:08:14] Max-Forwards: 70 [2023-11-23 19:08:14] From: "Unknown" ;tag=e687dae2-76dc-42e3-b5bb-4a8dec0b3e96 [2023-11-23 19:08:14] To: ;tag=qfkx79v12.iki. [2023-11-23 19:08:14] Call-ID: 4398c167-be94-4a19-b2c1-fffa24fa6e7c [2023-11-23 19:08:14] CSeq: 29990 INVITE [2023-11-23 19:08:14] Contact: [2023-11-23 19:08:14] Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE [2023-11-23 19:08:14] Allow-Events: talk,hold [2023-11-23 19:08:14] Require: 100rel [2023-11-23 19:08:14] RSeq: 39930 [2023-11-23 19:08:14] User-Agent: snomM400/06.70.0202 (MAC=000413D11E6B; SER= 00000; HW=3) [2023-11-23 19:08:14] Content-Length: 0 ``` Any idea that could help to unblock ? Thanks in advance for your help ! ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [November 23, 2023, 11:30pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/2 "2023-11-23T23:30:18Z")
I think answered calls always get inband progress. Os endpoints\_list degenerate (single contact from single endpoint)? If not, the remote ringing won’t be passed through, but r should still work Are there any commas between r and b? There should not be. ---
**Author:** ![Vince\_be](https://avatars.discourse-cdn.com/v4/letter/v/59ef9b/32.png) [@Vince\_be](https://community.asterisk.org/u/Vince_be)\ **Post date:** [November 24, 2023, 7:33am UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/3 "2023-11-24T07:33:00Z")
Hi David, Thanks for your answer. inband\_progress is set to no by default it seems ([https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample](https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample)) Here is the value of endpoints\_list=PJSIP/vielsalm-m400-1 (dynamic value, can have several endpoints linked to implementation of hotdesking). There is no commas between r & b… Perhaps I should check with provider if there is something specific at his side. Regards ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [November 24, 2023, 11:20am UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/4 "2023-11-24T11:20:08Z")
> [@Vince\_be](#): > > inband\_progress is set to no by default it seems I’m not sure my point was clear. Once you answer a call, there is no way, in SIP, of signalling ringback, other than as inband audio. ---
**Author:** ![Vince\_be](https://avatars.discourse-cdn.com/v4/letter/v/59ef9b/32.png) [@Vince\_be](https://community.asterisk.org/u/Vince_be)\ **Post date:** [November 24, 2023, 1:14pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/5 "2023-11-24T13:14:28Z")
I updated inband\_progress=yes for all endpoints but without success ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [November 24, 2023, 1:44pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/6 "2023-11-24T13:44:37Z")
As I said, that would only make any difference if you hadn’t answered the call. Once you answer the call Asterisk has to use inband regardless of the setting. Are you using a codec in pass through mode? Audio tones are always generated as signed linear, so have to be transcoded. ---
**Author:** ![Vince\_be](https://avatars.discourse-cdn.com/v4/letter/v/59ef9b/32.png) [@Vince\_be](https://community.asterisk.org/u/Vince_be)\ **Post date:** [November 24, 2023, 4:39pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/7 "2023-11-24T16:39:14Z")
Finally the solution is really… stupid. I decided to start from a “blank” installation without any configuration files and to add them only when needed. Problem is that I never added the file : indications.conf As soon as this one was added (copy/paste from sample folder), the ringback tone was available ! Thanks for your help david551 and sorry for the stupid mistake 😉 ---
**Author:** ![Vince\_be](https://avatars.discourse-cdn.com/v4/letter/v/59ef9b/32.png) [@Vince\_be](https://community.asterisk.org/u/Vince_be)\ **Post date:** [November 24, 2023, 5:13pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/8 "2023-11-24T17:13:37Z")
By the way, is it a good practice to limit as much as possible the number of configuration files ? For sure, it generates error messages when loading the system due to missing conf files… Here is my list of files : ![image](https://global.discourse-cdn.com/asterisk/original/2X/e/e619293d0b19a8fa026400d8a236992f97ee6806.png) Thanks in advance (and again) for valuable feedback ! ---
**Author:** ![Pooh](https://avatars.discourse-cdn.com/v4/letter/p/f05b48/32.png) [@Pooh](https://community.asterisk.org/u/Pooh)\ **Post date:** [November 24, 2023, 5:28pm UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/9 "2023-11-24T17:28:46Z")
> By the way, is it a good practice to limit as much as possible the number > of configuration files ? I would say the answer is exactly the same as general security advice regarding packages / applications / services on your machine - if you don’t want something to be running, don’t have it installed. > For sure, it generates error messages when loading the system due to > missing conf files… In that case I suspect you should also be editing /etc/asterisk/modules.conf and preventing the modules whose configuration files you do not need from trying to load. Antony. ---
**Author:** ![sedwards](https://avatars.discourse-cdn.com/v4/letter/s/838e76/32.png) [@sedwards](https://community.asterisk.org/u/sedwards)\ **Post date:** [November 25, 2023, 12:14am UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/10 "2023-11-25T00:14:35Z")
> [@Vince\_be](#): > > is it a good practice to limit as much as possible the number of configuration files I do. I start with ‘autoload = no’ in modules.conf and only load the modules I need. “Parts left out don’t get broke.” ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [December 25, 2023, 12:14am UTC](https://community.asterisk.org/t/no-ringback-for-incoming-calls/99771/11 "2023-12-25T00:14:46Z")
Closed on December 25, 2023, 12:14am UTC This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.