search for: speex32

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the

...'t use cdex here is my error log could vorbis be the colpret? Event Type: Error Event Source: Application Error Event Category: None Event ID: 1000 Date: 7/9/2003 Time: 3:53:16 PM User: N/A Computer: HOME-XS1NC5AM3V Description: Faulting application cdex.exe, version 1.0.0.1, faulting module speex32.acm, version 1.0.0.0, fault address 0x0000b1fa. For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. Data: 0000: 41 70 70 6c 69 63 61 74 Applicat 0008: 69 6f 6e 20 46 61 69 6c ion Fail 0010: 75 72 65 20 20 63 64 65 ure cde 0018: 78 2e 65 78 65 20 3...

...o slin16 : (alaw)->(slin)->(slin16) alaw To siren7 : No Translation Path alaw To siren14 : No Translation Path alaw To testlaw : (alaw)->(slin)->(testlaw) alaw To g719 : No Translation Path alaw To speex32 : (alaw)->(slin)->(slin32)->(speex32) alaw To slin12 : (alaw)->(slin)->(slin12) alaw To slin24 : (alaw)->(slin)->(slin24) alaw To slin32 : (alaw)->(slin)->(slin32) alaw To slin44 : (alaw)->(slin)...

...context=out-local disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001 aors=2001 direct_media=no rtp_symmetric=yes force_rport=yes allow=alaw allow=speex allow=speex16 allow=speex32 allow=gsm [2001] type=aor qualify_frequency=5000 authenticate_qualify=yes max_contacts=1 remove_existing=yes [2001] type=auth auth_type=userpass password=test username=test Best Regards, Madushan -------------- next...

...isk 11 (VM, Cloud or even physical machine). Is it slin?, adding this overhead or there is something I am overlooking?. * * *Asterisk 11.0.1 => core show translation **(in microseconds)* *gsm ulaw alaw g726 adpcm slin lpc10 g729 speex speex16 ilbc g726aal2 g722 slin16 testlaw speex32 slin12 slin24 slin32 slin44 slin48 slin96 slin192* *gsm *- 15000 *15000 *15000 15000 9000 15000 15000 *15000 *23000 15000 15000 17250 17000 15000 23000 17000 17000 17000 17000 17000 17000 17000 *ulaw *15000 - 9150 15000 15000 9000 15000 15000 15000 23000 15...

...transport=system-udp > auth=2001 > aors=2001 > direct_media=no > rtp_symmetric=yes > force_rport=yes > allow=alaw > allow=speex > allow=speex16 > allow=speex32 > allow=gsm > > > [2001] > type=aor > qualify_frequency=5000 > authenticate_qualify=yes > max_contacts=1 > remove_existing=yes > > [2001] > type=auth > auth...

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

...mat GSM for ID 3 Found audio description format G729 for ID 18 Found audio description 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|...

...format speex for ID 98 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 96 Capabilities: us - (gsm|ulaw|alaw|g729|g722), peer - audio=(ulaw|alaw|speex16|g722|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.2:10054 Looking for 3 in thorsten_sip_for_testing (domain myhost.org) list_...

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

...stered 'audio' codec 'speex' at sample rate '16000' with id '20' == Created cached format with name 'speex16' == Registered 'audio' codec 'speex' at sample rate '32000' with id '21' == Created cached format with name 'speex32' == Registered 'audio' codec 'ilbc' at sample rate '8000' with id '22' == Created cached format with name 'ilbc' == Registered 'audio' codec 'g722' at sample rate '16000' with id '23' == Created cached format with n...