12-28-2015 06:05 AM - edited 03-17-2019 05:20 AM
Hi guys,
Can anyone help my out with this.
Recently a company has changed from SIP provider X to SIP provider Y
Voice traffic is based on SIP without any cube.
SIP provider Y has provided me the signaling IP with port 5060
I have modified the dial-peers in that particular voicegateway, when i do a debug ccsip message i dont see any voice traffic.
In the dial-peers for incoming and outgoing calls i have changed the session target ip addres for the new SIP provider please dial-peers below:
However i dont see any voicetraffic on the voice.
Did i forget something, what else can i troubleshoot??
dial-peer voice 10012 voip
description **** company SIP provider outgoing international *****
translation-profile outgoing VoIP-Interoute-Outgoing
preference 1
destination-pattern 00T
session protocol sipv2
session target ipv4:195.81.174.10
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 11010 voip
description **** Company SIP provider incoming *****
translation-profile incoming VoIP-Interoute-Incoming
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:195.81.174.10
incoming called-number 9604361[012].
dtmf-relay rtp-nte
codec g711alaw
no vad
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 600 min 60
no update-callerid
sip-ua
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
sip-server ipv4:195.81.174.10:5060
connection-reuse
host-registrar
They are not registrar so no username or password is required.
XXXXXX ( VG#195.81.174.10 5060
Trying 195.81.174.10, 5060 ... Open
Any help would be appreciated.
12-28-2015 06:39 AM
Hi,
Can you do a shut / no shut on the dial-peers and check if any output is generated for "debug ccsip messages" for a test call?
Manish
12-28-2015 06:44 AM
Slightly confused here as you claim there is no CUBE, yet you provided CUBE configuration. Please provide more details on your topology.
Also, provide the output of the "debug ccsip messages" for outbound so we can at least see if there is anything going out, please also provide "debug voice ccapi inout"
12-28-2015 08:21 PM
I would step through each item here, placing a test call after each (with debug ccsip messages turned on) and see if you are able to see SIP messages.
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