This is one of those "why is there no sound?" questions. My setup is a
Cisco 2611 with NM-2V and VIC-2BRI-S/T-TE. I'm on a Euro-ISDN
(basic-net3) connection. The config's below (phone numbers changed). I've
also attached output of various diagnostic commands below, as well as some
comments at the bottom.
isdn switch-type basic-net3
!
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729br8
codec preference 3 g729r8
codec preference 4 g723r63
codec preference 5 g723r53
codec preference 6 g711ulaw
!
voice class codec 2
codec preference 1 gsmfr
codec preference 2 gsmefr
!
interface BRI1/0
no ip address
no ip route-cache
isdn switch-type basic-net3
isdn incoming-voice voice
isdn answer1 32580600
!
voice-port 1/0/0
compand-type a-law
cptone GB
!
dial-peer voice 2 pots
application session
incoming called-number 32580600
destination-pattern T
direct-inward-dial
port 1/0/0
!
dial-peer voice 1 voip
application session
destination-pattern 32580668
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
!
sip-ua
sip-server ipv4:192.168.1.106
!
192.168.1.106 is my laptop with Windows XP and a softphone. I've checked
that the softphone (both SJPhone and eStara) work fine by dialing
another Windows box with a softphone. Procedure:
1. I dial my mobile phone by entering sip:12345678@192.168.1.102
(12345678 represents my mobile phone number, and 192.168.1.102 is the
router's IP address).
2. The call goes through fine. However there's no sound, whether I
speak into the mobile phone or my laptop's microphone. Output of "show
isdn status":
ISDN BRI1/0 interface
dsl 1, interface ISDN Switchtype = basic-net3
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 105, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
1 Active Layer 3 Call(s)
CCB:callid=16, sapi=0, ces=1, B-chan=1, calltype=VOICE
Active dsl 1 CCBs = 1
The Free Channel Mask: 0x80000002
show voice call:
1/0/0 1
vtsp level 0 state = S_CONNECT
callid 0x8008 B01 state S_TSP_CONNECT clld 12345678 cllg
1/0/0 2 - - -
1/0/1 1 - - -
1/0/1 2 - - -
show voice dsp:
DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK
COUNT
==== === == ======== ======= ===== ======= === == ========= == =====
============
C542 001 01 gsmfr 4.1.38 busy idle 0 0 1/0/0.1 NA 0
13049/20747
show voice trace 1/0/0:
1/0/0 State Transitions: timestamp (state, event) -> (state, event) ...
8420.532 (S_SETUP_INDICATED, E_CC_PROCEEDING) ->
8420.904 (S_PROCEEDING, E_CC_ALERT) ->
8422.014 (S_ALERTING, E_CC_CONNECT) ->
show rtp call:
No Active calls found
show call active voice brief:
Total call-legs: 2
11FC : 842050hs.1 +151 pid:2 Answer 12345678 active
dur 00:02:42 tx:7209/237897 rx:4408/145464
Tele 1/0/0 (28) [1/0/0] tx:162700/44080/0ms gsmfr noise:-71 acom:45
i/0:-67/-76 dBm
11FC : 842053hs.1 +145 pid:1 Originate 32580600 active
dur 00:02:42 tx:4408/145464 rx:7209/237897
IP 192.168.1.106:16384 rtt:0ms pl:10380/17600ms lost:903/1392/1219
delay:70/70/210ms gsmfr
Note that I have one voice-class with G.7xx codecs and another with GSM
codecs. I've tried both. I've also tried SJPhone on Linux.
Could the problem be related to the fact that there's no RTP
connection going on? What's weird is, if I dial digits on my softphone
(I guess DTMF or RFC2833), I can actually here the DTMF tones on my
mobile phone, but not the other way around. I guess this is because
the DTMF is transferred over RFC2833 and outputted by the Cisco box.
Another thing that's weird is the "i/0 -67/-76 dBm" -- what are normal
values for these two figures?
I hope that someone can help.
Guan
PS. I've previously posted this message at comp.dcom.sys.cisco with
slightly different output.
Cisco 2611 with NM-2V and VIC-2BRI-S/T-TE. I'm on a Euro-ISDN
(basic-net3) connection. The config's below (phone numbers changed). I've
also attached output of various diagnostic commands below, as well as some
comments at the bottom.
isdn switch-type basic-net3
!
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729br8
codec preference 3 g729r8
codec preference 4 g723r63
codec preference 5 g723r53
codec preference 6 g711ulaw
!
voice class codec 2
codec preference 1 gsmfr
codec preference 2 gsmefr
!
interface BRI1/0
no ip address
no ip route-cache
isdn switch-type basic-net3
isdn incoming-voice voice
isdn answer1 32580600
!
voice-port 1/0/0
compand-type a-law
cptone GB
!
dial-peer voice 2 pots
application session
incoming called-number 32580600
destination-pattern T
direct-inward-dial
port 1/0/0
!
dial-peer voice 1 voip
application session
destination-pattern 32580668
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
!
sip-ua
sip-server ipv4:192.168.1.106
!
192.168.1.106 is my laptop with Windows XP and a softphone. I've checked
that the softphone (both SJPhone and eStara) work fine by dialing
another Windows box with a softphone. Procedure:
1. I dial my mobile phone by entering sip:12345678@192.168.1.102
(12345678 represents my mobile phone number, and 192.168.1.102 is the
router's IP address).
2. The call goes through fine. However there's no sound, whether I
speak into the mobile phone or my laptop's microphone. Output of "show
isdn status":
ISDN BRI1/0 interface
dsl 1, interface ISDN Switchtype = basic-net3
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 105, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
1 Active Layer 3 Call(s)
CCB:callid=16, sapi=0, ces=1, B-chan=1, calltype=VOICE
Active dsl 1 CCBs = 1
The Free Channel Mask: 0x80000002
show voice call:
1/0/0 1
vtsp level 0 state = S_CONNECT
callid 0x8008 B01 state S_TSP_CONNECT clld 12345678 cllg
1/0/0 2 - - -
1/0/1 1 - - -
1/0/1 2 - - -
show voice dsp:
DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK
COUNT
==== === == ======== ======= ===== ======= === == ========= == =====
============
C542 001 01 gsmfr 4.1.38 busy idle 0 0 1/0/0.1 NA 0
13049/20747
show voice trace 1/0/0:
1/0/0 State Transitions: timestamp (state, event) -> (state, event) ...
8420.532 (S_SETUP_INDICATED, E_CC_PROCEEDING) ->
8420.904 (S_PROCEEDING, E_CC_ALERT) ->
8422.014 (S_ALERTING, E_CC_CONNECT) ->
show rtp call:
No Active calls found
show call active voice brief:
Total call-legs: 2
11FC : 842050hs.1 +151 pid:2 Answer 12345678 active
dur 00:02:42 tx:7209/237897 rx:4408/145464
Tele 1/0/0 (28) [1/0/0] tx:162700/44080/0ms gsmfr noise:-71 acom:45
i/0:-67/-76 dBm
11FC : 842053hs.1 +145 pid:1 Originate 32580600 active
dur 00:02:42 tx:4408/145464 rx:7209/237897
IP 192.168.1.106:16384 rtt:0ms pl:10380/17600ms lost:903/1392/1219
delay:70/70/210ms gsmfr
Note that I have one voice-class with G.7xx codecs and another with GSM
codecs. I've tried both. I've also tried SJPhone on Linux.
Could the problem be related to the fact that there's no RTP
connection going on? What's weird is, if I dial digits on my softphone
(I guess DTMF or RFC2833), I can actually here the DTMF tones on my
mobile phone, but not the other way around. I guess this is because
the DTMF is transferred over RFC2833 and outputted by the Cisco box.
Another thing that's weird is the "i/0 -67/-76 dBm" -- what are normal
values for these two figures?
I hope that someone can help.
Guan
PS. I've previously posted this message at comp.dcom.sys.cisco with
slightly different output.