pjsip | PJSIP is Open Source SIP , Media , and NAT Traversal Library | TCP library

I am looking to replace my older Asterisk 15 VoIP Server with Asterisk 17. Currently I have an application that works with Asterisk over ARI (i.e. Asterisk Rest Interface) + WS (for events). On my current version everything works fine, but after conncting my app to Asterisk 17 it did not receive ChannelHangupRequest event and the following events in time (i.e. only after ~30 seconds). So I've decided to take a look on the Asterisk logs and I've found out that:

1. The problem is not caused by my app
2. The ChannelHangupRequest event is send, as I said before, after about 30-50 seconds. The interesting part is that it's cause is equal to 1 and there is a field soft equal to true. You can see below the event in JSON format. Official Asterisk 17 documentation does not give an obvious description for it or at least it is not obvious to me: From soft: boolean (optional) - Whether the hangup request was a soft hangup request. Also, official hangup cause mappings indicate that 1 is mapped to AST_CAUSE_UNALLOCATED. On my older version (i.e. Asterisk 15), where the event fires immediately, case is equal to 16 = AST_CAUSE_NORMAL_CLEARING.

<{"cause":1,"soft":true,"type":"ChannelHangupRequest","timestamp":"2021-03-23T17:10:19.817+0000","channel":{"id":"1616519370.38","name":"PJSIP/1000-00000026","state":"Up","caller":{"name":"Cristi","number":"1000"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-internal","exten":"s","priority":1,"app_name":"Stasis","app_data":"my-app"},"creationtime":"2021-03-23T17:09:30.364+0000","language":"en"},"asterisk_id":"02:42:ac:13:00:03","application":"my-app"} < {"type":"StasisEnd","timestamp":"2021-03-23T17:10:19.817+0000","channel":{"id":"1616519370.38","name":"PJSIP/1000-00000026","state":"Up","caller":{"name":"Cristi","number":"1000"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-internal","exten":"s","priority":1,"app_name":"Stasis","app_data":"my-app"},"creationtime":"2021-03-23T17:09:30.364+0000","language":"en"},"asterisk_id":"02:42:ac:13:00:03","application":"my-app"}

1. Asterisk log shows that channel was disconnected for lack of audio RTP activity in 49 seconds. It explains the behaviour to some extent, specifically, it explains where the ChannelHangupRequest event comes from (after ~50 seconds), but it does not explain why isn't it sent properly immediately after user hangups.

13373[2021-03-23 17:09:28] VERBOSE[6273] stasis/app.c: Creating Stasis app 'my-app' 13374[2021-03-23 17:09:28] VERBOSE[6273] res_http_websocket.c: WebSocket connection from '172.19.0.1:54308' for protocol '' accepted using version '13' 13375[2021-03-23 17:09:29] VERBOSE[6276] stasis/app.c: Replacing Stasis app 'my-app' 13376[2021-03-23 17:09:29] VERBOSE[6276] res_http_websocket.c: WebSocket connection from '172.19.0.1:54312' for protocol '' accepted using version '13' 13377[2021-03-23 17:09:30] VERBOSE[6278] stasis/app.c: Replacing Stasis app 'my-app' 13378[2021-03-23 17:09:30] VERBOSE[6278] res_http_websocket.c: WebSocket connection from '172.19.0.1:54322' for protocol '' accepted using version '13' 13379[2021-03-23 17:09:30] VERBOSE[6280] dial.c: Called 1000 13380[2021-03-23 17:09:30] VERBOSE[1601] netsock2.c: Using SIP RTP Audio TOS bits 184 13381[2021-03-23 17:09:30] VERBOSE[1601] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field. 13382[2021-03-23 17:09:30] VERBOSE[1601] netsock2.c: Using SIP RTP Audio CoS mark 5 13383[2021-03-23 17:09:30] VERBOSE[6280] dial.c: PJSIP/1000-00000026 is ringing 13384[2021-03-23 17:09:30] VERBOSE[6280] dial.c: PJSIP/1000-00000026 is ringing 13385[2021-03-23 17:09:44] VERBOSE[6276] res_http_websocket.c: WebSocket connection from '172.19.0.1:54312' closed 13386[2021-03-23 17:09:45] VERBOSE[6282] stasis/app.c: Replacing Stasis app 'my-app' 13387[2021-03-23 17:09:45] VERBOSE[6282] res_http_websocket.c: WebSocket connection from '172.19.0.1:54334' for protocol '' accepted using version '13' 13388[2021-03-23 17:09:49] VERBOSE[1601] res_rtp_asterisk.c: 0x7f8c1c02bf90 -- Strict RTP learning after remote address set to: 192.168.10.172:10096 13389[2021-03-23 17:09:49] VERBOSE[6280] dial.c: PJSIP/1000-00000026 answered 13390[2021-03-23 17:09:49] WARNING[1601] channel.c: Unable to find a codec translation path: (ulaw) -> (g723) 13391[2021-03-23 17:09:49] WARNING[1601] channel.c: Unable to find a codec translation path: (g723) -> (ulaw) 13392[2021-03-23 17:09:49] VERBOSE[6280] ari/resource_channels.c: Launching Stasis(my-app) on PJSIP/1000-00000026 13393[2021-03-23 17:10:19] NOTICE[1108] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/1000-00000026' for lack of audio RTP activity in 49 seconds

1. The above described behaviour applies only when user answers and, then, drops. If the user rejects call without answering, everything works fine.

One solution would be to reduce the RTP inactivity timeout as a workaround, but this wouldn't be preferable. Do you have any suggestions?

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PJSIP - 2.9

I am trying to enable ALAW and ULAW code into my iOS app. I passed list of codec for account add with G711A/8000/1 and G711U/8000/1 but when I make call and check the INVTE I did not see anyone of codec there.

Whereas if I add another codec I can see that enabled in INVITE. I tried to check whether these codecs has some other dependency or it has be enabled like G729.

Please help me if someone can. Thank you

I am trying to make a simple SIP user agent using https://github.com/pjsip/pjproject. I can succesfully connect to a sip server (Twilio) and place calls to PSTN numbers using the pjsua_* interface. This works fine.

What I would like now is to get a notification (through a callback or such) from pjsip when the user that I am calling answers the call.

I am using on_call_state() to get updates on the invite, but this goes through the same states

CALLING -> CONNECTING -> CONFIRMED -> DISCONNCTD

even if the user rejects the call. So I guess I am not looking at the right callback for this.

How can I definitely tell if the user has answered or rejected the call?

I have been going through Android Pjsip pjsua2 sample app. It works but they didn't implemented registration listener. And couldn't find any good enough documentation as well. Eg for android native sip stack, they have registration listener for registration failure, registration success as well. Please help me with a sample code as well if its possible.

I have two different set of json object as follows.

I am struggling with this : when entering a queue, I'd like to play a welcome message to the caller.

The caller enters first a short queue (initQ) without announce, then enters the mynewQ where I'd like to play a custom welcome message.

So far, I was able to play only the default queue-youarenext.alaw with :

Files queues.conf :

I am trying to wrap my head around ARI and Asterisk, my goal is to dial from an extension to another. I dialed 5001 from extension 5002. Now in the stasisStart function, I want to create a new channel, and used the dialed number (5001) and pass 'PJSIP/5001' to the endpoint. How do I get the dialed number?

Dialplan: