02-01-2015 08:12 AM - edited 03-17-2019 01:48 AM
I have an new SIP trunk set on an UC520 and the incoming calls are ok, but the outgoing calls are getting an busy tone(not working).
The bellow trace is showing that the cause is "No route to destination (3) ". The question is this route has to do with the firewall(ip routing) or with the voice translation rules?
001866: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x843CE50C
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 0777777777 <- main sip number
Called Number : 0888888888 <- called number
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port : 0.0.0.0:0
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name :
001867: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 3
Disconnect Cause (SIP) : 200
02-01-2015 11:40 AM
Please try to specify interface from where SIP call will go
voice service voip
sip
bind control source-interface "VOIP_INTERFACE"
bind media source-interface "VOIP_INTERFACE"
02-01-2015 10:13 PM
Thanks I will give it a go
02-03-2015 12:30 AM
Please debug call
debug ccsip calls
May be it is because of wrong dial-peers
Also some providers ask to send Caller-ID as Outside number ( I mean that calling outside You should configure your City number as outgoing)
02-03-2015 12:50 AM
Hi Emil,
The provider is providing full number caller id, the outgoing number is in form of 0111111111. The user to make an call is dialing 9 then 0222222222 the external number
Bellow is the debug calls and sh run
UC_520#debug ccsip calls
SIP Call statistics tracing is enabled
UC_520#
002875: //14407/5E8706E1BE27/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x843D0BD8
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 0111111111
Called Number : 0222222222
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port : 0.0.0.0:0
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name :
002876: //14407/5E8706E1BE27/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 3
Disconnect Cause (SIP) : 200
02-03-2015 12:52 AM
Please remove
voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME
02-03-2015 12:55 AM
this too
voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL
02-03-2015 01:05 AM
Hi Emil,
I've removed both, but no change.
Almost all the config is made through CCA.
02-03-2015 11:47 PM
The issue was resolved in part on the provider's side.
On UC520 side by adding an outbound rule to fa0/0, to allow traffic to SIP IP. And adding an static route to allow 0.0.0.0/0.0.0.0 IP default gateway.
02-03-2015 12:21 AM
Hi Emil,
I've added bind control and media interface but outgoing calls are the same blocked, strange thing is that the cause is still no route to destination (3)
but
UC_520#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP User Agent bind status(media): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl
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