06-26-2015 04:30 AM - edited 03-17-2019 03:29 AM
Hello,
I have a Cisco 1760 with 6 digital noIP phones and one connection to the LAN. There is a SIP connection to a Cloud SIP provider.
When a phone makes a call, after a correct SIP negotiation, the rtp traffic just comes from the external called phone (the external SIP Proxy), but no rtp traffic is generated from the 1760. So the receiver can speak but can´t listen.
I show you the config. The SIP connection is on dial-peer voice 100, the others are phones.
Any advice will be appreciated!
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname TL_PBX
!
boot-start-marker
boot-end-marker
!
no aaa new-model
clock timezone GMT 1
voice-card 1
!
voice-card 2
!
voice-card 3
!
!
!
no ip routing
no ip cef
!
!
ip name-server 8.8.4.4
ip name-server 8.8.8.8
!
multilink bundle-name authenticated
!
!
!
voice service voip
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
!
!
!
!
voice translation-rule 1
rule 1 /0*/ //
!
!
voice translation-profile DiscardDigit0
translate called 1
!
voice translation-profile Local-CLID
translate calling 8
!
!
!
!
archive
log config
hidekeys
!
!
!
translation-rule 1
!
gw-accounting aaa
!
!
!
!
!
interface FastEthernet0/0
ip address dhcp
no ip route-cache
speed auto
full-duplex
!
ip default-gateway 192.168.1.254
!
!
ip http server
ip http authentication local
ip http secure-server
!
!
!
!
!
tftp-server flash:P00405000700.bin
tftp-server flash:P00405000700.sbn
!
control-plane
!
!
!
voice-port 1/0
!
voice-port 1/1
!
voice-port 2/0
!
voice-port 2/1
!
voice-port 3/0
!
voice-port 3/1
!
!
!
!
!
!
dial-peer voice 100 voip
tone ringback alert-no-PI
description SIP trunk to S
translation-profile outgoing DiscardDigit0
service session
destination-pattern 0T
session protocol sipv2
session target ipv4:sip_ip
session transport udp
codec g711alaw
no vad
!
dial-peer voice 1 pots
destination-pattern 201
port 2/0
!
dial-peer voice 2 pots
destination-pattern 202
port 2/1
!
dial-peer voice 3 pots
destination-pattern 203
port 3/0
!
dial-peer voice 4 pots
service session
destination-pattern 204
port 3/1
!
dial-peer voice 5 pots
destination-pattern 205
port 1/0
!
dial-peer voice 6 pots
destination-pattern 206
port 1/1
!
sip-ua
credentials username --
authentication username --
nat symmetric check-media-src
registrar dns:-- expires 3600
sip-server ipv4:sipserver
!
!
06-26-2015 05:21 AM
Hello,
Take a look on this docs:
http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/5219-fix-1way-voice.html
Regards
Leonardo Santana
06-30-2015 10:49 AM
Hello,
thank you for your responses.
In the voice-card config, the commands dsp or dspfarm are not available, but i think the dsp is enabled generally:
show dspfarm
DSPFARM Configuration Information:
Admin State: UP, Oper Status: ACTIVE - Cause code: NONE
Transcoding Sessions: 2(Avail: 2), Conferencing Sessions: 0 (Avail: 0)
Trans sessions for mixed-mode conf: 0 (Avail: 0), RTP Timeout: 100
Connection check interval 600 Codec G729 VAD: ENABLED
I copy the shows you asked for when a call is done:
sh voice dsp group all (nothing)
but I made a show voice dsp before the call...
sh voice dsp
DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT
==== === == ======== ========== ===== ======= === == ========= == ===== ============
C549 000 00 g711ulaw 9.4.0 Idle Idle 0 0 1/0 NA 0 1480/1451
C549 000 01 g711ulaw 9.4.0 Idle Idle 0 0 1/1 NA 0 9/3
C549 001 00 g711ulaw 9.4.0 Idle Idle 0 0 2/0 NA 0 1075/1035
C549 001 01 g711ulaw 9.4.0 Idle Idle 0 0 2/1 NA 0 9/3
Active Voice Call details
Current total analog signalling channels: 4
Current max allowed digital timeslot for voice: 0
Current active transcoding sessions: 0
Current free transcoding sessions: 2
Current active conferencing sessions: 0
Current free conferencing sessions: 0
Current number of DSP group: 1
Group 0:
Current allocated analog signalling channels: 4
Current free analog signalling channels: 0
Current allocated digital signalling channels: 0
Current free digital signalling channels: 0
Port(s) served: 1/0 1/1 2/0 2/1
Current Available MIPS: 300
SPMM DSPRM State Image D-sig D-sig A-sig A-sig Mips Voice/Xcode
Dsp Dsp allocate free allocate free Free Chan
0/0 0 UP FIXHC 0 0 2 0 100 0
0/1 1 UP FIXHC 0 0 2 0 100 0
0/2 2 UP XCODE 0 0 0 0 100 0
show call active voice brief
<ID>: <CallID> <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>
long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops>
speeds(bps): local <rx>/<tx> remote <rx>/<tx>
Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
bw: <req>/<act> codec: <audio>/<video>
tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Media call-legs: 0
Total call-legs: 2
1201 : 225 9565370ms.1 +23240 pid:5 Answer 205 active
dur 00:00:43 tx:2276/382368 rx:2275/364000
Tele 1/0 (225) [1/0] tx:45510/45510/0ms g711alaw noise:-26 acom:15 i/0:-22/-46 dBm
1201 : 226 9583020ms.1 +5590 pid:100 Originate 679984243 active
dur 00:00:43 tx:2275/0 rx:231/4169624
IP 14.1.5.8:13570 SRTP: off rtt:0ms pl:2000/40ms lost:0/0/0 delay:65/65/65ms g711alaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Media call-legs: 0
Total call-legs: 2
show voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 226 225 17600 13570 192.168.201.102 194.140.135.80
Found 1 active RTP connections
Thank you!
06-26-2015 06:06 AM
Hi,
I don't see DSP services enabled on your router
voice-card 0
dspfarm
dsp services dspfarm
During the call, please share the following output.
sh voice dsp group all
sh call active voice brief
sh voip rtp connection
07-02-2015 02:18 PM
j.sarceda,
you can't use ip default gateway as route to outside. use static route to gateway which DHCP provide you. so your packets can go to outside. to verify this I suppose you can't get reply message if you pinging your router from outside. check that
regards
please rate if it's helpful
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