Firstly, Thank you SO MUCH for your thoughtful and comprehensive response. Your info was so helpful to me, and to others I’m sure. I have read the links you sent and made the suggested changes. My codec was the g729 but I don’t need it for any particular reason so I changed it (see below).
QUOTE: "Most gateways do not require SIP authentication. Can you leave everything blank in the SIP account except the name? If not you want to build backwards for a minimum config."
RESPONSE: I tried to take out everything but the Router FXO interface requires a un/pw combo. So I left it 1001/1001 but there is an option below that says “Call without Registration:” And I set it to “enable” from the default disable. I assume this means “don’t send un/pw”. What do you think?
In the suggested trunk settings from your previous post: should there be a “context=from-pstn” or something like that included?
My Versions:
(From FreePBX 2.8.1.5) configuring Asterisk (Ver. 1.8.11.0)
In Asterisk SIP settings, should NAT settings be set to yes, no, never or route? I am using NAT for internet access, but my Asterisk server is on same physical network as router and my router FXO card is sending out calls through PSTN - So I have this set to “no” currently.
I have Public IP set, since I do not route VoIP traffic past my network (PSTN connection only). Is this correct?
since I am using Grandstream GXP2110 phones & my FXO card only supports g726, g729, g723.1, g711u/a, I have non-standard g726 set to “yes” and in my trunk settings I put in “allow=g726”. My phones only support PCMU/A, G722, G723.1, G726-32, iLBC. I removed ULAW from option because it didn’t look like it was installed? (see below codec chart)
On voip-info.org I read “For Grandstream phones: set dtmfmode=info” And “dtmfmode=info ; either RFC2833 or INFO for the BudgeTone” So should I change this in trunk settings? What about on my FXO port config? Currently it’s set to rfc2883 both in asterisk and FXO port.
Here are my codecs from asterisk:
Translation times between formats (in microseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw
g723 - - - - - - - - - - - - - - - - - - -
gsm - - 3001 3002 8999 3001 3000 8999 - 25997 28996 8999 4999 - - 8998 - 25995 3001
ulaw - 5001 - 1 6000 2 1 6000 - 22998 25997 6000 2000 - - 5999 - 22996 2
alaw - 5001 1 - 6000 2 1 6000 - 22998 25997 6000 2000 - - 5999 - 22996 2
g726aal2 - 7999 3000 3001 - 3000 2999 8998 - 25996 28995 8998 4998 - - 8997 - 25994 3000
adpcm - 6000 1001 1002 6999 - 1000 6999 - 23997 26996 6999 2999 - - 6998 - 23995 1001
slin - 5000 1 2 5999 1 - 5999 - 22997 25996 5999 1999 - - 5998 - 22995 1
lpc10 - 9999 5000 5001 10998 5000 4999 - - 27996 30995 10998 6998 - - 10997 - 27994 5000
g729 - - - - - - - - - - - - - - - - - - -
speex - 9000 4001 4002 9999 4001 4000 9999 - - 29996 9999 5999 - - 9998 - 26995 4001
ilbc - 8999 4000 4001 9998 4000 3999 9998 - 26996 - 9998 5998 - - 9997 - 26994 4000
g726 - 7000 2001 2002 7999 2001 2000 7999 - 24997 27996 - 3999 - - 7998 - 24995 2001
g722 - 8999 4000 4001 9998 4000 3999 9998 - 26996 29995 9998 - - - 3999 - 20996 4000
siren7 - - - - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - - - - -
slin16 - 12998 7999 8000 13997 7999 7998 13997 - 30995 33994 13997 3999 - - - - 16997 7999
g719 - - - - - - - - - - - - - - - - - - -
speex16 - 16998 11999 12000 17997 11999 11998 17997 - 34995 37994 17997 7999 - - 4000 - - 11999
testlaw - 5001 2 3 6000 2 1 6000 - 22998 25997 6000 2000 - - 5999 - 22996 -
and here is my log from asterisk full:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK04db3ee0
From: "Unknown" <sip:[email protected]>;tag=as0f06f185
To: <sip:10.10.10.1>;tag=2287315620
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
content-type: application/sdp
Allow: INVITE, ACK, INFO, OPTIONS, NOTIFY, REFER, CANCEL, BYE
Content-Length: 272
v=0
o=username 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 0 8 18 4 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jul 8 16:44:27] VERBOSE[5662] chan_sip.c: --- (9 headers 13 lines) ---
[Jul 8 16:44:27] VERBOSE[5662] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[Jul 8 16:44:28] VERBOSE[5645] config.c: == Parsing '/etc/asterisk/res_pktccops.conf': [Jul 8 16:44:28] VERBOSE[5645] config.c: == Found
[Jul 8 16:45:01] VERBOSE[5630] asterisk.c: -- Remote UNIX connection
[Jul 8 16:45:01] VERBOSE[28902] asterisk.c: -- Remote UNIX connection disconnected
[Jul 8 16:45:27] VERBOSE[5662] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.1:5060:
OPTIONS sip:10.10.10.1 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK6754512a
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as1936f60f
To: <sip:10.10.10.1>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Sun, 08 Jul 2012 21:45:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Jul 8 16:45:27] VERBOSE[5662] chan_sip.c:
<--- SIP read from UDP:10.10.10.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK6754512a
From: "Unknown" <sip:[email protected]>;tag=as1936f60f
To: <sip:10.10.10.1>;tag=120317160
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
content-type: application/sdp
Allow: INVITE, ACK, INFO, OPTIONS, NOTIFY, REFER, CANCEL, BYE
Content-Length: 272
v=0
o=username 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 0 8 18 4 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jul 8 16:45:27] VERBOSE[5662] chan_sip.c: --- (9 headers 13 lines) ---
[Jul 8 16:45:27] VERBOSE[5662] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[Jul 8 16:46:27] VERBOSE[5662] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.1:5060:
OPTIONS sip:10.10.10.1 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK754a2ddd
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as68004d73
To: <sip:10.10.10.1>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Sun, 08 Jul 2012 21:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Jul 8 16:46:27] NOTICE[5662] chan_sip.c: Registration from '<sip:[email protected]>' failed for '10.10.10.41:5060' - No matching peer found
[Jul 8 16:46:27] VERBOSE[5662] chan_sip.c:
<--- SIP read from UDP:10.10.10.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK754a2ddd
From: "Unknown" <sip:[email protected]>;tag=as68004d73
To: <sip:10.10.10.1>;tag=2543903276
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
content-type: application/sdp
Allow: INVITE, ACK, INFO, OPTIONS, NOTIFY, REFER, CANCEL, BYE
Content-Length: 272
v=0
o=username 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 0 8 18 4 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jul 8 16:46:27] VERBOSE[5662] chan_sip.c: --- (9 headers 13 lines) ---
[Jul 8 16:46:27] VERBOSE[5662] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[Jul 8 16:47:27] VERBOSE[5662] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.1:5060:
OPTIONS sip:10.10.10.1 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK08e751b7
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as3024c902
To: <sip:10.10.10.1>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Sun, 08 Jul 2012 21:47:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Jul 8 16:47:28] VERBOSE[5662] chan_sip.c:
<--- SIP read from UDP:10.10.10.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK08e751b7
From: "Unknown" <sip:[email protected]>;tag=as3024c902
To: <sip:10.10.10.1>;tag=458416752
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
content-type: application/sdp
Allow: INVITE, ACK, INFO, OPTIONS, NOTIFY, REFER, CANCEL, BYE
Content-Length: 272
v=0
o=username 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 0 8 18 4 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jul 8 16:47:28] VERBOSE[5662] chan_sip.c: --- (9 headers 13 lines) ---
[Jul 8 16:47:28] VERBOSE[5662] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[Jul 8 16:48:28] VERBOSE[5662] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.1:5060:
OPTIONS sip:10.10.10.1 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK7efffa21
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as67ff091c
To: <sip:10.10.10.1>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Sun, 08 Jul 2012 21:48:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Jul 8 16:48:28] VERBOSE[5662] chan_sip.c:
<--- SIP read from UDP:10.10.10.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.4:5060;branch=z9hG4bK7efffa21
From: "Unknown" <sip:[email protected]>;tag=as67ff091c
To: <sip:10.10.10.1>;tag=3281003956
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
content-type: application/sdp
Allow: INVITE, ACK, INFO, OPTIONS, NOTIFY, REFER, CANCEL, BYE
Content-Length: 272
v=0
o=username 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 0 8 18 4 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jul 8 16:48:28] VERBOSE[5662] chan_sip.c: --- (9 headers 13 lines) ---
[Jul 8 16:48:28] VERBOSE[5662] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[Jul 8 16:48:51] NOTICE[5650] chan_iax2.c: No registration for peer '7777' (from 127.0.0.1)
[Jul 8 16:48:51] NOTICE[5649] chan_iax2.c: No registration for peer '7777' (from 127.0.0.1)
[Jul 8 16:49:04] VERBOSE[5662] netsock2.c: == Using SIP RTP TOS bits 184
[Jul 8 16:49:04] VERBOSE[5662] netsock2.c: == Using SIP RTP CoS mark 5
[Jul 8 16:49:04] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:1] ResetCDR("SIP/763-0000001a", "") in new stack
[Jul 8 16:49:04] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:2] NoCDR("SIP/763-0000001a", "") in new stack
[Jul 8 16:49:04] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:3] Progress("SIP/763-0000001a", "") in new stack
[Jul 8 16:49:04] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:4] Wait("SIP/763-0000001a", "1") in new stack
[Jul 8 16:49:05] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:5] Progress("SIP/763-0000001a", "") in new stack
[Jul 8 16:49:05] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:6] Playback("SIP/763-0000001a", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[Jul 8 16:49:05] VERBOSE[28915] file.c: -- <SIP/763-0000001a> Playing 'silence/1.gsm' (language 'en')
[Jul 8 16:49:06] VERBOSE[28915] file.c: -- <SIP/763-0000001a> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[Jul 8 16:49:11] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:7] Wait("SIP/763-0000001a", "1") in new stack
[Jul 8 16:49:11] NOTICE[5662] chan_sip.c: Registration from '<sip:[email protected]>' failed for '10.10.10.23:5060' - No matching peer found
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [2143845635@from-internal:8] Congestion("SIP/763-0000001a", "20") in new stack
[Jul 8 16:49:12] WARNING[28915] channel.c: Prodding channel 'SIP/763-0000001a' failed
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: == Spawn extension (from-internal, 2143845635, 8) exited non-zero on 'SIP/763-0000001a'
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [h@from-internal:1] Macro("SIP/763-0000001a", "hangupcall") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/763-0000001a", "1?endmixmoncheck") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Goto (macro-hangupcall,s,9)
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:9] NoOp("SIP/763-0000001a", "End of MIXMON check") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/763-0000001a", "1?nomeetmemon") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Goto (macro-hangupcall,s,15)
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:15] NoOp("SIP/763-0000001a", "MEETME_RECORDINGFILE=") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:16] GotoIf("SIP/763-0000001a", "1?noautomon") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Goto (macro-hangupcall,s,18)
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:18] NoOp("SIP/763-0000001a", "TOUCH_MONITOR_OUTPUT=") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:19] GotoIf("SIP/763-0000001a", "1?noautomon2") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Goto (macro-hangupcall,s,25)
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:25] NoOp("SIP/763-0000001a", "MONITOR_FILENAME=") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:26] GotoIf("SIP/763-0000001a", "1?skiprg") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Goto (macro-hangupcall,s,29)
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:29] GotoIf("SIP/763-0000001a", "1?skipblkvm") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Goto (macro-hangupcall,s,32)
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:32] GotoIf("SIP/763-0000001a", "1?theend") in new stack
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Goto (macro-hangupcall,s,34)
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: -- Executing [s@macro-hangupcall:34] Hangup("SIP/763-0000001a", "") in new stack
[Jul 8 16:49:12] VERBOSE[28915] app_macro.c: == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/763-0000001a' in macro 'hangupcall'
[Jul 8 16:49:12] VERBOSE[28915] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/763-0000001a'
I feel like I am closer since at least the message changed when I dial - now it’s “Your call can’t be completed as dialed, please check number and dial again” (then gives busy tones).
Suggestions? My guess is that it’s a connection issue with the trunk FXO port?
What did you mean in your response when you said "Most gateways do not require SIP authentication. Can you leave everything blank in the SIP account except the name? If not you want to build backwards for a minimum config."
What is it to “build backwards for a minimum config”?