Hi everyone, I have a problem with calls in PJSIP . When I freshly lunch my application the calls works perfectly then after 2 hours I can’t makes calls anymore and I had this message. 17:21:01.617 pjsua_aud.c .Set sound device: capture=-1, playback=-2 17:21:01.617 pjsua_aud.c …Opening sound device (speaker + mic) PCM@16000/1/20ms 17:21:01.634 ec0x629fc888 …AEC created, clock_r...
Pjsip2.8 is tested with "pjsip-app/bin/pjsua-mipsel-unknown-elf" and the following error is generated during the call. Alsa_dev.c ca_thread_func: overrun! Alsa_dev.c pb_thread_func: underrun! Cpu occupancy reached 96%. I can provide ssh to let you log in remotely. Codes is wm8960，Alsa drive is ok, I can play music with madplay Can anyone help me find
Hello Everyone, I need somebody who can compile again PJSIP library for PJSUA with all support library similar to following link with latest OpenSSL library. CSipSimple Sample: [войдите, чтобы посмотреть URL] Work expected: - Make necessary changes to successfully compile the PJSIP source code that is available in the above link. -
Help me configure [войдите, чтобы посмотреть URL] and my Twilio account to send and receive phone calls usi...Console. Final solution should be able to call PSTN from Raspberry Pi Command line using PJSUA. PJSIP and PJSUA are already compiled and installed on Pi, and sound card should be working, so all that is needed now is proper configuration of Twilio and PJSUA
Hello, I am searching a freelancer who has experience in PjSIP/PjSUA "[войдите, чтобы посмотреть URL]". Who can help me compile my source pjsip.c for "BusyBox v1.01" so I can easily run it directly in "Busybox terminal". The binary must contain all the libraries in it. Thank you
...use a service to run the PJSIP component. Communication between the app and the service will be through Intents and the service will not be bound. The service can use PJSUA or some User Agent equivalent. The application can be developed using some publically available service such as Asterisk or Callcentric, but would have to use our custom
...will use a service to run the PJSIP component. Communication between the app and the service will be through Intents and the service will not be bound. The service can use PJSUA or some User Agent equivalent. The application can be developed using some publically available service such as Asterisk or Callcentric, but would have to use our custom SIP
I need an Asterisk expert to install ...setup asterisk Test it using: [войдите, чтобы посмотреть URL] Once I verify, I will release the funds thanks Links [войдите, чтобы посмотреть URL] https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support [войдите, чтобы посмотреть URL]
I need someone to troubleshoot an a...asterisk Test it using: [войдите, чтобы посмотреть URL] Once I verify, I will release the funds Thanks. Links I used: [войдите, чтобы посмотреть URL] https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support [войдите, чтобы посмотреть URL]
I saw your post for I need a compiled version of pjproject-2.0-alpha2 with all C# sample codes. The source code is available at: [войдите, чтобы посмотреть URL] I need the same thing but I need this to be a B2BUA where I can take in a call and do some business logic then fork the call to my providers. I want to make a Switch for dialer traffic and
Microsip Project is written in c + +, and using the library PJSUA. microsip Open source SIP softphone for Windows based on PJSIP stack = [войдите, чтобы посмотреть URL] Lib. PJSUA = [войдите, чтобы посмотреть URL] What to do: complete rebranding and change a little bit design
... Creation Time Description Resource Path Location Type 1318048279562 undefined reference to `pjsip_tls_transport_start' pjsua_core.c /VoipProject_StarTeleLogic/pjsip/src/pjsua-lib line 1598 C/C++ Problem How to resolve this and how to enable TLS ? Note : I am following [войдите, чтобы посмотреть URL] Its so urgent work so
We would like a modified simple-pjsua with build-in tone detection routine implemented eg. Goertzel algorithm The application should compile on linux systems For testing purpose the aplication should use the sip registration details received from the command line eg. dtmf-test sip_domain sip_user sip_pass The uri called will always respond
...konsolowa, ja potrzebuję podobną funkcjonalność ale w wersji okienkowej. Jest dostępny pełny kod źródłowy z komentarzami, dokumentacja, API. Konkretnie chodzi o program PJSUA opisany na tej stronie: [войдите, чтобы посмотреть URL] Docelowy wygląd programu: [войдите, чтобы посмотреть URL] To
We have already a fully running python app which uses pjsua and need someone to add 2 things for us: The budget is 45$ and it is only the extension. Delivery time: 1-2 days MAX. You do NOT have to develop something from scratch - only extend the current version, please. 0. The current version plays an audio like an answering machine and allows user
...listening to info audio; # for disconnecting this call; or 0 to repeat the intro." 2. next, it decides on the DTRM signal that the caller pressed how to go on: The pjsip/pjsua provides an event for doing so: on_dtmf_digit. depending on the input (pressed key/DTRM), it goes on with playing [войдите, чтобы посмотреть URL], repeating the [войдите, чтобы посмотреть URL] or disconnectin...
I would like someone to write a C++ wraper on top of the pjsip library. The code should wrap pjsip library (not pjsua). the Object/functionality I need: 1. Account - account will be responsible for registration. No support for stun is needed, how ever, rport does need support. also support for UDP/TCP transport is needed. 2. Call
...Za ogólny wzór można wziąć program typu ZoIPer. Do tworzenia tego typu programów jest dostępna darmowa biblioteka - PJSIP, [войдите, чтобы посмотреть URL] wraz z przykładem implementacji - PJSUA, więc nie trzeba samemu odkrywać koła, tylko złożyć program z gotowych klocków. Aplikacja ma zawierać podstawowe funkcje spotykane w tego typu programach: - moż...
I am looking for consultence about using pjsip. what i need is to use pjsua for the signaling, this part is already covered. the project is about getting hooks on the basic rtp channel so that i can use the channel without using the hole pjmedia lib.
...the maximum parallel calls provided by the library from 32 to a higher level possibly to 50 or more. We could do this with the old library by modifying *pjsip/include/pjsua-lib/pjsua.h* and increasing PJSUA_MAX_CALLS and PJSUA_MAX_PLAYERS and compile it that way in the old library there was a call which could return the compiled maximum
Our existing VOIP/SIP conferencing server has to be completely rewritten from scratch. The current application is built using PJSUA 0.8 with the Python binding. The new application has to be built on PJSUA 1.0 (if you know other open source library which does the same you are free to come forward with recommendations, but it has to be
Looking for a RTP/SIP stack that can connect to media ports such as sound cards and microphones. The requirements are very similar to pjlib which includes the pjmedia, pjlib, pjsua. There should be a low layer library that can implement the protocols and should be multithread friendly so both servers and clients can be built. A high level library should