Archives : January-2018
users-list-mail-to-blog, my name is Mary and im from Russia. Currently I live in US. Im so glad to see your profile on Facebook. You seem like my type and I would like us to know each other better. You are super cute and handsome. If you feel the sa..
users-list-mail-to-blog, the hottest man in the world! My name is Katya and im from Russia, but currently I live in the USA. I just wanted to let you know that I liked you from your photos and would like to know more about you. Let me know if you wo..
So as yall know, with your help I managed to get Opus installed at last. Yay!With excitement, I wrote my dialplan, dialled in, and….[Jan 28 21:30:11] ERROR[29977][C-0000001d]: format_ogg_opus.c:95ogg_opus_rewrite: Cannot write OGG/Opus streams. So..
Before I got an log a ticket, can I just check Im not doing anything wrong?In 15.2, to install Opus:1) run `make menuselect`2) Highlight Codec Translators and press enter.3) Scroll down to codec_opus in the section labeled External4) Press enter to sel..
All, Running asterisk 11.25.3, the /proc/cpuinfo says Intel(R) Xeon(R) CPU X5675@ 3.07GHzlsmod | grep dahdi gives dahdi_transcode163841 wctc4xxp dahdi_voicebus 614402 wctdm24xxp,wcte12xp dahdi 22528011wctdm24xxp,wcfxo,wctdm,dahdi_transcode,oct612x dahdi_voicebus,wcb4xxp,wct1xxp,wct4xxp,wcte11xp,wcte1..
i met with this interesting situation [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport 8 since no request was received in 32 seconds [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting d..
I have an old setup based in Asterisk 1.8. The carrier is accepting Ulaw in the initial invite, but immediately the call is established they send a re-invite to change to Alaw. This doesnt get transcoded and the user gets no audio from after the re-inv..
All;I had someone ask me if they received an incoming phone call and it was forwarded off pbx to their cell phone, would the call be strictly between the caller and the cell phone, or would it between the caller, the pbx, and the cell phone where ..
I want to start recording with a prompt of press or say 1 to 5. If no DMTF is pressed, I want to send the recording to Google Speech to get the number back (got that part working already).If any dtmf key is pressed while Application_Recordis runn..
I know that hangup handlers (https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers) have to finish quickly.So its no surprise that my speech to text agi which takes 8 seconds gets killed.However, can anyone think of a way round this? So, once ..