Hello.After successful compilation 15.6.1 (bundled pjsip) and start asterisk i has error Symbol pjsip_tls_transport_start2 not found./main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 and pjsip_tls_transport_start.More:* All versi..
Author : Dmitriy Serov
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Asterisk 15.2.0, pjsip Yesterday I installed 15.3.0-rc1 (after 15.2.0). Today I had to rollback to 15.2.0 (not 15.2.2). The reason is: the loaded server very often hung on the same places: SUBSCRIBE/NOTIFY processing. Unloading modules was the solut..
[sub-out-do-dial] exten => s,1,NoOp(Dial) same => n,NoOp(FirstChannel: ${CHANNEL}) same => n,Dial(????,60,gF) same => n,NoOp(SecondChannel: ${CHANNEL}) same => n,Return() [some] exten => s,1,GoSub(sub-out-do-dial,s,1) In case of the destinat..
Hello.Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs.Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5…
There is a separate app for recording voice (app_record) or dtmf input (app_read). But there is no way to allow the user to choose to enter by voice or by keypad in same time.app_record analyzes the dtmf input, but only the # and * (to quit). Noth..
asterisk 13.8.7, PJSIP.One VoIP provider requires a specific value in the field contact of a INVITE.What setting does indicate the value will be in this field (instead asterisk)?Thanks.currect settings (with templates):[srv_d22778](srv-auth)username00999x..
At the moment I plan to migrate from asterisk 13.7 to 13.8. Because of relatively frequent updates I am building asterisk from a directory that is updated via git switch to the desired branch.1. Will be updated pjproject patches with git pull?2. W..
Good day.Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device.Is suspect two things:1. del..
Please help find the cause of strange behavior res_pjsip.Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up!Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4..
Good day.I have a problem when using android native sip client. When dialplan used Progress (sending 183 Session Progress) after some seconds android native sip client declines a call (the logs are at the end of). No ealry media be heard.In same c..