search for: aal2

...answers to my earlier posting. I hope I have better luck this time- The question: Is it possible to use Asterisk as media gateway controller? I know * supports MGCP, but does that also imply that I can use * to control any third party media gateway (such as one providing media conversion from E1 to AAL2). - DL

...t=0 0 > m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32 > inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:99 G726-32/8000 > a=rtpmap:108 AAL2-G726-32/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > a=direction:active > a=nortpproxy:yes > <-------------> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21 l...

...=root 2136927789 2136927789 IN IP4 192.168.2.28 > s=call > c=IN IP4 123.231.72.210 > t=0 0 > m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:99 G726-32/8000 > a=rtpmap:112 AAL2-G726-32/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > <--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878 > <http://123.231.72.210:33878> ---> > SIP/...

Hi, I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

...(0x2) audio gsm (GSM) 4 (1 << 2) (0x4) audio ulaw (G.711 u-law) 8 (1 << 3) (0x8) audio alaw (G.711 A-law) 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 << 5) (0x20) audio adpcm (ADPCM) 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 << 7) (0x80) audio lpc10 (LPC10)...

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

...SM) >> 4 (1 << 2) (0x4) audio ulaw >> (G.711 u-law) >> 8 (1 << 3) (0x8) audio alaw >> (G.711 A-law) >> 16 (1 << 4) (0x10) audio g726aal2 >> (G.726 AAL2) >> 32 (1 << 5) (0x20) audio adpcm >> (ADPCM) >> 64 (1 << 6) (0x40) audio slin (16 >> bit Signed Linear PCM) >> 128 (1 << 7)...

...guration parameter called 'g726nonstandard' has been added to sip.conf, so that Asterisk can use the packing order expected by the device (even though it requested a different order). In addition, the internal format number for G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The result of this is that this version of Asterisk will be able to interoperate over IAX2 with older versions of Asterisk, as long as this version is told to allow 'g726aal2' instead of 'g726' as the codec for the call. Now, I can complete a call but I have now audio yet....

...(0x1) audio g723 (G.723.1) 2 (1 << 1) (0x2) audio gsm (GSM) 4 (1 << 2) (0x4) audio ulaw (G.711 u-law) 8 (1 << 3) (0x8) audio alaw (G.711 A-law) 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 << 5) (0x20) audio adpcm (ADPCM) 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 << 7) (0x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.72...

...(0x1) audio g723 (G.723.1) 2 (1 << 1) (0x2) audio gsm (GSM) 4 (1 << 2) (0x4) audio ulaw (G.711 u-law) 8 (1 << 3) (0x8) audio alaw (G.711 A-law) 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 << 5) (0x20) audio adpcm (ADPCM) 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 << 7) (0x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.72...

...nding directly to it by IP address). Thanks, Michelle ---------------------------------------------------------------------------- ------------ Audio is at 99.99.26.93 port 16738 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 100.100.116.29:5060: INVITE sip:112345991313483...

...78:abcd::2 14385 typ host a=candidate:Hdccea0f2 2 UDP 2130706430 2001:123:ab:123::2 14385 typ host a=candidate:Hcbb5ed22 2 UDP 2130706430 fe80::21f:c6ff:fec4:926a 14385 typ host a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrat...

Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I feel it's just a configuration issue but I just can't get my head wrapped around it. The situation is basically this: I have an Asterisk connected to an Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no dedicated hardware

...d port 47732 [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x4 (ulaw) to SDP [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x2 (gsm) to SDP [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x8 (alaw) to SDP [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x10 (g726aal2) to SDP [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x20 (adpcm) to SDP [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x40 (slin) to SDP [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x80 (lpc10) to SDP [Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x400 (i...

...CE[4997]: core.cpp:2133 sfa_call_hangup: ending call following are output of some commands:- *CLI> core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723 - - - - - - - - - - - - - gsm - - 2 2 2 2 1 2 6 - - 2 - ulaw - 2 - 1 2 2 1 2 6 - - 2 - alaw - 2...

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

...ile adding to bridge (Reported by Ilya Trikoz) * ASTERISK-24900 - Manager event ParkedCallSwap is not documented (Reported by Rusty Newton) * ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator (Reported by Corey Farrell) * ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when negotiating g.726 (Reported by Kevin Harwell) * ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first dialed party (Reported by Janusz Karolak) * ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer call started from Macro (Reported by Arveno Santo...

...my $call_type = "VoIP"; $call_type = "Telephony" if $event{'Channel'} =~ /^(Zap)|(VPB)|(phone)|(Modem)|(CAPI)|(mISDN)|(Console)/; # session-protocol # other, cisco, h323, multicast, sipv2, sdp, frf11-trunk, cisco-switched, MarsAnalog, C1000Isdn, aal2-trunk my $protocol = 'other'; $protocol = 'sipv2' if $event{'Channel'} =~ /^SIP/i; $protocol = 'h323' if $event{'Channel'} =~ /^h323/i; $channels{$event{'Channel'}} = { 'CHANNEL' => $event{'Channel'}, 'CALL...

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