search for: silk16

...asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from <zoiperipaddr>: > requested format = speex, > requested prefs = (), > actual format = ulaw, > host prefs = (silk16|ulaw|gsm|g722), > priority = mine -- Executing [8447 at nz-in:1] Dial("IAX2/n4-270", "IAX2/sydney") in new stack -- Called IAX2/sydney -- Call accepted by <nyipaddr> (format ulaw) -- Format for call is (ulaw) -- IAX2/sydney-8819 is ring...

...0 Calltoken req: No Trunk : No Encryption : No Callerid : "" <> Expire : -1 ACL : No Addr->IP : (Unspecified) Port 0 Defaddr->IP : 0.0.0.0 Port 4569 Username : Codecs : (gsm|ulaw|g722) Codec Order : (silk16|ulaw|gsm|g722) Status : UNKNOWN Qualify : every 60000ms when OK, every 10000ms when UNREACHABLE (sample smoothing Off) iax.conf: [general] bandwidth=medium trunkmtu=1240 disallow=all allow=silk16 allow=ulaw allow=gsm allow=g722 jitterbuffer=yes forcejitterbuffer=no authdebug=y...

On 11.9.0: > -- Accepting AUTHENTICATED call from 111.xxx.yyy.zzz: > -- > requested format = speex, > -- > requested prefs = (), > -- > actual format = ulaw, > -- > host prefs = (silk16|ulaw|gsm|g722), > -- > priority = mine > -- Executing [8447 at voip-in:1] Dial("IAX2/n4-5734", "IAX2/ncal") in new stack > -- Called IAX2/ncal > -- Call accepted by 68.xxx.yyy.zzz (format ulaw) > -- Format for call is (ulaw) >...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

...iption format telephone-event for ID 101 Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 66.241.99.161:11460 <--- SIP read from UDP:64....

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

...egistered 'audio' codec 'silk' at sample rate '12000' with id '41' == Created cached format with name 'silk12' == Registered 'audio' codec 'silk' at sample rate '16000' with id '42' == Created cached format with name 'silk16' == Registered 'audio' codec 'silk' at sample rate '24000' with id '43' == Created cached format with name 'silk24' == Sorcery registered wizard 'bucket' == Sorcery registered wizard 'bucket_file' == Parsing '/etc/asterisk/so...

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc