Missing asterisk directives.

[pw44]

Hya,

installing my sip line (voipcheap), i did note it did not register.
So, looking into the database, table ast_config, i noted that some directives were missing.

Code: [Select]

83 0 18 0 sip.conf general alwaysauthreject yes
85 0 18 0 sip.conf general nat                        yes
86 0 60 0 sip.conf general externhost        mydyndns.homeunix.org
87 0 5 0 sip.conf general externrefresh        5
88 0 60 0 sip.conf general localnet               192.168.80.0/255.255.255.0
89 0 9 0 sip.conf general allow                g729
90 0 10 0 sip.conf general allow                g723
91 0 101 0 sip.conf general register                pwollny:XXXXXXXXX@sip.voipcheap.com/2062036594
The 83 is very inportant for fail2ban and 85-88 to have the NAT.
I also would like to know which are the rules for cat_metric and var_metric.

Anyway, my sip line register, but i keep getting

Code: [Select]

> doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
in the asterisk verbose, i can place calls, but no sound incoming and outgoing and receiving a call is rejected with "no service message" in my asterisk.

Way to solve it?

TIA

[cfernandes]

Paulo ,

for test can you enable  ALLOW_SIP_ANON = yes

Carlos

[pw44]

Hi Carlos,
i will test and report.
Best regards,
Paulo

[pw44]

Hi Carlos,
i did set ALLOW_SIP_ANON = yes,
When i call from my mobile to my DID which routes to voipcheap, my phone rings. When i pick up, i talk and can hear in my mobile.
When i talk in the mobile i CANNOT hear in my home phone.
Calling from my home phone to my mobile, nothing happens.
So, calling from outside to home works, and only sound out, no sound in.
Calling out does not work.
Did you test voipcheap?
TIA,
Paulo

[pw44]

Hya,

installing my sip line (voipcheap), i did note it did not register.
So, looking into the database, table ast_config, i noted that some directives were missing.

Code: [Select]

83 0 18 0 sip.conf general alwaysauthreject yes
85 0 18 0 sip.conf general nat                        yes
86 0 60 0 sip.conf general externhost        mydyndns.homeunix.org
87 0 5 0 sip.conf general externrefresh        5
88 0 60 0 sip.conf general localnet               192.168.80.0/255.255.255.0
89 0 9 0 sip.conf general allow                g729
90 0 10 0 sip.conf general allow                g723
91 0 101 0 sip.conf general register                pwollny:XXXXXXXXX@sip.voipcheap.com/2062036594
The 83 is very inportant for fail2ban and 85-88 to have the NAT.
I also would like to know which are the rules for cat_metric and var_metric.

Anyway, my sip line register, but i keep getting

Code: [Select]

> doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
in the asterisk verbose, i can place calls, but no sound incoming and outgoing and receiving a call is rejected with "no service message" in my asterisk.

Way to solve it?

TIA

After updating asterisk, all the entries i did were gone
No more NAT, externip, etc....