Unable to create channel of type 'SIP'

Asterisk Asterisk Support
1

Hi
I am getting this error, I haven’t modified anything but only network team has changed firewall settings. It was working fine before that.

Could anyone tell me the cause for this error?

the full log…

[Jul 18 03:24:28] NOTICE[6658] chan_sip.c: Peer '601' is now UNREACHABLE! Last qualify: 0 [Jul 18 03:24:28] NOTICE[6658] chan_sip.c: Peer '201' is now UNREACHABLE! Last qualify: 0 [Jul 18 03:24:51] VERBOSE[6666] logger.c: -- Executing [601@default:1] Dial("SIP/201-00000000", "SIP/601|60|") in new stack [Jul 18 03:24:51] WARNING[6666] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Jul 18 03:24:51] VERBOSE[6666] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Jul 18 03:24:51] VERBOSE[6666] logger.c: -- Executing [601@default:2] Goto("SIP/201-00000000", "default|85026666666666601|1") in new stack [Jul 18 03:24:51] VERBOSE[6666] logger.c: -- Goto (default,85026666666666601,1) [Jul 18 03:24:51] VERBOSE[6666] logger.c: -- Executing [85026666666666601@default:1] Wait("SIP/201-00000000", "1") in new stack [Jul 18 03:24:52] VERBOSE[6666] logger.c: -- Executing [85026666666666601@default:2] VoiceMail("SIP/201-00000000", "601|u") in new stack [Jul 18 03:24:52] VERBOSE[6666] logger.c: -- <SIP/201-00000000> Playing 'vm-theperson' (language 'en') [Jul 18 03:24:53] VERBOSE[6666] logger.c: -- <SIP/201-00000000> Playing 'digits/6' (language 'en') [Jul 18 03:24:54] VERBOSE[6666] logger.c: -- <SIP/201-00000000> Playing 'digits/0' (language 'en') [Jul 18 03:24:55] VERBOSE[6666] logger.c: -- <SIP/201-00000000> Playing 'digits/1' (language 'en') [Jul 18 03:24:55] VERBOSE[6666] logger.c: -- <SIP/201-00000000> Playing 'vm-isunavail' (language 'en') [Jul 18 03:24:57] VERBOSE[6666] logger.c: -- <SIP/201-00000000> Playing 'vm-intro' (language 'en')

sip show peers;

Name/username              Host               Dyn Nat ACL Port     Status
601/601                    <myIP>                 D   N      15802    UNREACHABLE
201/201                    <myIP                   D   N      5060     UNREACHABLE

network team has given that they see below error

Attack Malformed SIP datagram 
Attack Information Invalid or no 'CSEQ' field

601 is logged on using XLite and 201 is logged in using LinPhone.

Using XLite, everytime it is showing different port here… does it really use these ports instead of 5060, this is another question I have.

Thanks in advance.

2

Your enpoints need to be allowed to reach your asterisk server on these ports (default):
udp/5060
udp/10000 to udp/20000

Plus, it seems your firewall has some ALG or some kind of DPI, which is most certainly causing issue.

Regards,

Constantin

3

As well as the phones not being regisrtered, the syntax used went out with Asterisk 1.4.