<div dir="ltr"><div class="gmail_quote">On Tue, Jul 19, 2011 at 10:12 PM, Matthew J. Roth <span dir="ltr"><<a href="mailto:mroth@imminc.com">mroth@imminc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="im">Michael wrote:<br>
><br>
> True. In the working system, LAN calls are also using G.729, while<br>
> in the non-working system, LAN calls are in G.711 (supported but<br>
> not prioritized by the phones) and only the SIP trunk to the ITSP<br>
> is set to G.729.<br>
<br>
</div>Can you set the phone to G.711 and try making a LAN call on the non-<br>
working system. If a call that is G.711 from end-to-end doesn't have<br>
the same problem it would be evidence of a codec translation issue.<br></blockquote><div>That's the full trace of a call in G.711:<br><br>Reliably Transmitting (no NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a>:<br>
INVITE <a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4751ec69;rport<br>Max-Forwards: 70<br>From: "9000" <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
To: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>Contact: <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>><br>Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>
CSeq: 102 INVITE<br>User-Agent: FPBX-2.8.1(1.6.2.19)<br>Date: Thu, 21 Jul 2011 07:12:31 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 261<br><br>v=0<br>o=root 1017783905 1017783905 IN IP4 192.168.1.10<br>s=Asterisk PBX 1.6.2.19<br>c=IN IP4 192.168.1.10<br>t=0 0<br>m=audio 17696 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br>---<br> -- Called 500<br><br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>SIP/2.0 180 Ringing<br>
From: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>To: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 102 INVITE<br>Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69<br>Supported: replaces,100rel<br>
User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>Content-Length: 0<br><br><br><-------------><br>--- (10 headers 0 lines) ---<br> -- SIP/500-00000330 is ringing<br>
<br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>SIP/2.0 200 OK<br>From: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
To: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>
CSeq: 102 INVITE<br>Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69<br>Supported: replaces,100rel<br>User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE<br>Content-Type: application/sdp<br>Content-Length: 226<br><br>v=0<br>o=500 1096000000 0 IN IP4 192.168.1.109<br>s=Audio Session<br>i=Audio Session<br>c=IN IP4 192.168.1.109<br>
t=0 0<br>m=audio 16384 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=ptime:20<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-11<br><br><-------------><br>--- (12 headers 11 lines) ---<br>Found RTP audio format 0<br>
Found RTP audio format 101<br>Found audio description format PCMU for ID 0<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>Peer audio RTP is at port <a href="http://192.168.1.109:16384">192.168.1.109:16384</a><br>list_route: hop: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>
set_destination: Parsing <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>> for address/port to send to<br>set_destination: set destination to 192.168.1.109, port 5060<br><br>Transmitting (no NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a>:<br>
ACK <a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1265e92e;rport<br>Max-Forwards: 70<br>From: "9000" <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
To: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>Contact: <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>><br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 102 ACK<br>User-Agent: FPBX-2.8.1(1.6.2.19)<br>Content-Length: 0<br><br><br>---<br> -- SIP/500-00000330 answered SIP/Smile-0000032f<br>
[Jul 21 10:12:36] WARNING[13622]: dsp.c:1360 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833<br><br><br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>
INVITE <a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a> SIP/2.0<br>From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>
To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>
CSeq: 1 INVITE<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c04d6-373f24ae<br>Max-Forwards: 70<br>Supported: replaces,100rel<br>User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE<br>Content-Type: application/sdp<br>Content-Length: 294<br><br>v=0<br>o=500 1096000001 0 IN IP4 192.168.1.109<br>s=SIPPhone Session<br>i=Audio Session<br>c=IN IP4 0.0.0.0<br>
t=0 0<br>m=audio 16384 RTP/AVP 0 8 18 101<br>a=sendonly<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-11<br><br><-------------><br>
--- (13 headers 14 lines) ---<br>Sending to 192.168.1.109 : 5060 (no NAT)<br><br><--- Transmitting (no NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c04d6-373f24ae;received=192.168.1.109<br>
From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 1 INVITE<br>Server: FPBX-2.8.1(1.6.2.19)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>><br>Content-Length: 0<br><br><br><------------><br>Audio is at 192.168.1.10 port 17696<br>Adding codec 0x4 (ulaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br><br><--- Reliably Transmitting (no NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c04d6-373f24ae;received=192.168.1.109<br>
From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 1 INVITE<br>Server: FPBX-2.8.1(1.6.2.19)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>><br>Content-Type: application/sdp<br>Content-Length: 237<br><br>v=0<br>o=root 1017783905 1017783905 IN IP4 192.168.1.10<br>
s=Asterisk PBX 1.6.2.19<br>c=IN IP4 192.168.1.10<br>t=0 0<br>m=audio 17696 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br><------------><br>
<br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>ACK <a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a> SIP/2.0<br>From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>
To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>
CSeq: 1 ACK<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c0521-2b8a87aa<br>Max-Forwards: 70<br>User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>
Content-Length: 0<br><br><br><-------------><br>--- (10 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>INVITE <a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a> SIP/2.0<br>
From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 2 INVITE<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-321-c3cbc-39b225e8<br>Max-Forwards: 70<br>
Supported: replaces,100rel<br>User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE<br>Content-Type: application/sdp<br>
Content-Length: 300<br><br>v=0<br>o=500 1096000000 0 IN IP4 192.168.1.109<br>s=SIPPhone Session<br>i=Audio Session<br>c=IN IP4 192.168.1.109<br>t=0 0<br>m=audio 16384 RTP/AVP 0 8 18 101<br>a=sendrecv<br>a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-11<br><br><-------------><br>--- (13 headers 14 lines) ---<br>Sending to 192.168.1.109 : 5060 (no NAT)<br>
<br><--- Transmitting (no NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-321-c3cbc-39b225e8;received=192.168.1.109<br>
From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 2 INVITE<br>Server: FPBX-2.8.1(1.6.2.19)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>><br>Content-Length: 0<br><br><br><------------><br>Audio is at 192.168.1.10 port 17696<br>Adding codec 0x4 (ulaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br><br><--- Reliably Transmitting (no NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-321-c3cbc-39b225e8;received=192.168.1.109<br>
From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 2 INVITE<br>Server: FPBX-2.8.1(1.6.2.19)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>><br>Content-Type: application/sdp<br>Content-Length: 237<br><br>v=0<br>o=root 1017783905 1017783905 IN IP4 192.168.1.10<br>
s=Asterisk PBX 1.6.2.19<br>c=IN IP4 192.168.1.10<br>t=0 0<br>m=audio 17696 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br><------------><br>
<br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>ACK <a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a> SIP/2.0<br>From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>
To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>
CSeq: 2 ACK<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-322-c3ce9-34d41f51<br>Max-Forwards: 70<br>User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>
Content-Length: 0<br><br><br><-------------><br>--- (10 headers 0 lines) ---<br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>BYE <a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a> SIP/2.0<br>
From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>
Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>CSeq: 3 BYE<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-32b-c625a-41626b8<br>Max-Forwards: 70<br>
Supported: replaces,100rel<br>User-Agent: SIP Phone<br>Content-Length: 0<br><br><br><-------------><br>--- (10 headers 0 lines) ---<br>Sending to 192.168.1.109 : 5060 (no NAT)<br><br><--- Transmitting (no NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-32b-c625a-41626b8;received=192.168.1.109<br>From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe<br>
To: "9000"<<a href="mailto:sip%3A9000@192.168.1.10">sip:9000@192.168.1.10</a>>;tag=as40d1788b<br>Call-ID: <a href="mailto:3b81833312062ead03c47919333e549c@192.168.1.10">3b81833312062ead03c47919333e549c@192.168.1.10</a><br>
CSeq: 3 BYE<br>Server: FPBX-2.8.1(1.6.2.19)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Length: 0<br><br><br><------------><br><br><br>The same thing happens (or doesn't happen). MOH is not initiated when sendonly is received by Asterisk. I'm not an expert on this point, but I suspect that it's a system parameter somewhere and not related to codecs.<br>
</div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="im"><br>
> We tested with NAT set to "no" and "yes" and neither settings<br>
> mattered.<br>
<br>
</div>As long as the phone and the Asterisk server are both on the same LAN,<br>
my recommendation would be to test with NAT set to "no". NAT is not<br>
necessary unless there is a firewall between the phone and the<br>
Asterisk server and setting it to "no" also eliminates a variable that<br>
differentiates it from the working system.<br></blockquote><div><br>We changed it to "no". It doesn't matter.<br></div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br>
> It should (have modules loaded for both formats). How do we check<br>
> this?<br>
<br>
The following command should output a line for each module (as shown):<br>
<br>
# asterisk -rx 'module show' | egrep 'format_g729.so|format_pcm.so'<br>
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 5<br>
format_g729.so Raw G729 data 0<br></blockquote><div><br>That's what I get:<br>[root@pbx ~]# asterisk -rx 'module show' | egrep 'format_g729.so|format_pcm.so'<br>
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 <br>format_g729.so Raw G729 data 0 <br><br>What does the 5 (or in my case 0) stand for? <br>
</div><br>Thanks.<br></div></div>