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cisco 1760 SIP one direction sound

j.sarceda
Level 1
Level 1

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
!
!

4 Replies 4

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!

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

saif musa
Level 4
Level 4

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

 

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