A SIP AG is a voice gateway that exchanges SIP signals with other devices between the PSTN/ISDN and IP network. It can implement VoIP functions.
The packet switched network (PSN) development brings revolutionary changes to the voice phone system and many new technologies emerge. VoIP transmits voice services such as telephony services on an IP network, and the IP multimedia subsystem (IMS) promotes development of VoIP applications. An IMS network is a standard next generation carrier network that provides mobile or fixed-line multimedia services. It supports traditional packet switched and circuit switched telephony systems. Compared with the public switched telephone network (PSTN), VoIP features higher resource utilization and VoIP calls do not occupy telephone lines exclusively. VoIP has been applied for commercial use. The line switched PSTN has developed for many years and currently there are a large number of devices. Replacing the PSTN with VoIP takes high costs. A device can function as the SIP AG to connect the PS****o IP networks with low costs.
Please provide the below information for analysis this issue .
1-Display diagnostic-information from AR
2-Please specify two User examples facing this issue of forwarding calls
3-Below debugging is required when Issue happens
1 debugging information collect
[Huawei-diagnose]debugging voice sipmsg
[Huawei-diagnose]quit
[Huawei]quit
<Huawei>terminal debugging
<Huawei>terminal monitor
------ after the log is collected------
<Huawei>undo terminal debugging
<Huawei>undo terminal monitor
2 Please confirm the following configuration information: example
#
trunk-group CTGTrunk sip trunk-circuit
outgoing-right all
signalling-address ip 172.16.0.33 port 5063
media-ip 172.16.0.33
peer-address static 1.59.2.24 5060
register-uri xxxxxx /// Please confirm that this parameter is the same as that provided by the operator.
home-domain CTMEoffice
trunk-sipat0 CTMEA default-called-telno 00971 /// Please confirm that this parameter is the same as that provided by the operator.
#
trunk-group London sip no-register
enterprise CTMEA dn-set CTMEA_DXB
outgoing-right idd enable
signalling-address ip 172.16.0.33 port 5060
media-ip 172.16.0.33
peer-address static 95.xx.194.38 5060
home-domain London-sip-trunk
maxcr 10
#
The SIP trunk is the upper layer voice service based on the data link. By default, the SIP trunk is responsible for the interaction of UDP packets, including SIP signaling packets and RTP media packets. SIP trunk messages can be replaced by NAT addresses.


