Placing a call to a GSM gateway on different SIP port-1.8.3

Asterisk Asterisk Support
1

Hello

I just encountered a problem with Asterisk 1.8.3 when I try dialing out from a SIP Trunk on different port then 5060 (I tried 5064).

I have a SunComm GSM-VoIP gateway that I connect to Asterisk via SIP Trunk. The gateway has 8 SIM slots, and if I send a SIP Invite to port 5060 (from Asterisk to GSM-VoIP gateway for outgoing GSM call), the gateway automatically decides which SIM card it would choose. If I want to manually choose to which SIM card I want to send the call to, I need to send a SIP Invite to a pre-defined port (5064 for SIM1, 5066 for SIM2, 5068 for SIM3, …) on the SIP Trunk.

I want to send a call to the SIM1 of the GSM-VoIP gateway with the following command:

Test_GSM_Trunk is the name of the peer that is defined in the sip.conf.

This worked perfectly in Asterisk 1.6.x, but if I try this in 1.8.3, I get the following error:

[Mar 30 19:17:23] ERROR[6280]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("Test_GSM_Trunk", "5064", ...): Name or service not known
[Mar 30 19:17:23] WARNING[6280]: chan_sip.c:5057 create_addr: No such host: Test_GSM_Trunk:5064
[Mar 30 19:17:23] WARNING[6280]: acl.c:698 ast_ouraddrfor: Cannot connect
[Mar 30 19:17:23] WARNING[6280]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0xcc2fe38 (len 865) to (null) returned -1: Invalid argument
    -- Called Test_GSM_Trunk:5064/0000xxxxxx
[Mar 30 19:17:24] WARNING[2786]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0xcc2fe38 (len 865) to (null) returned -1: Invalid argument
[Mar 30 19:17:25] WARNING[2786]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0xcc2fe38 (len 865) to (null) returned -1: Invalid argument
[Mar 30 19:17:27] WARNING[2786]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0xcc2fe38 (len 865) to (null) returned -1: Invalid argument

I was checking the web if the Dial() application sytax has changed on Asterisk 1.8.3, but found no good info. Has anyone got any idea what am I doing wrong here?

2

No. “Test_GSM_Trunk:5064” is the name of a peer that is not in sip.conf and might not comply with Asterisk peer naming rules. The port number is a property of the peer and cannot be overidden in an address.

3

Whel, the Dial command in the first post worked perfectly on 1.6, so apparently this is a new rule in 1.8. So there is no way that I could make outgoing calls on different SIP ports, I have to create 8 peers each with different port?