8did.com provider

Have any of you used this provider before?
can you share the configuration you used on your asterisk?

I see nothing special in the description they published on http://8did.com/en/features#voip

To forward calls through an SIP device, you must provide an SIP address instead of a forwarding phone number. An SIP address is like an email address, some of these are: 8885551212@, 8885551212@mysip.mydomain.com, or even joe@mysip.mydomain.com.

This means that you should have a peer with host= matching their IP and a route for the number you’re forwarding to (8885551212 in the example above).

thanks i tried that configuration and all i got was
Got SIP response 480 “Temporarily Unavailable” back from

Please post your trunk configuration and a complete SIP trace of the call.

that is exactly the problem, i don’t have a trunk configuration for 8did

i got this trunk configuration fron an 8did.com representative:


and here is a cli capture of the call:

– Called SIP/Voicebuy/447874400728
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_START’,{ts ‘2016-11-22 09:12:09.4732’},’’,’’,’’,’’,’’,‘s’,‘to-voicebuy’,‘SIP/Voicebuy-0000012e’,’’,’’,3,’’,‘1479827529.302’,‘1479827528.301’,’’,’’,’’)]
– Got SIP response 500 “Server Internal Error” back from
– SIP/Voicebuy-0000012e is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:23] NoOp(“SIP/002-0000012d”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 38”) in new stack
– Executing [s@macro-dialout-trunk:24] GotoIf(“SIP/002-0000012d”, “0?continue,1:s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“SIP/002-0000012d”, “RC=38”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/002-0000012d”, “38,1”) in new stack
– Goto (macro-dialout-trunk,38,1)
– Executing [38@macro-dialout-trunk:1] Goto(“SIP/002-0000012d”, “continue,1”) in new stack
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘HANGUP’,{ts ‘2016-11-22 09:12:50.397469’},‘CID:15878850151’,‘7447874400728’,’’,’’,’’,‘7447874400728’,‘to-voicebuy’,‘SIP/Voicebuy-0000012e’,‘AppDial’,’(Outgoing Line)’,3,’’,‘1479827529.302’,‘1479827528.301’,’’,’’,’’)]
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] NoOp(“SIP/002-0000012d”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 38 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:2] Set(“SIP/002-0000012d”, “CALLERID(number)=002”) in new stack
– Executing [7447874400728@from-internal:6] Macro(“SIP/002-0000012d”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/002-0000012d”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/002-0000012d”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/002-0000012d”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/002-0000012d”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
– <SIP/002-0000012d> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_END’,{ts ‘2016-11-22 09:12:50.397520’},‘CID:15878850151’,‘7447874400728’,’’,’’,’’,‘7447874400728’,‘to-voicebuy’,‘SIP/Voicebuy-0000012e’,‘AppDial’,’(Outgoing Line)’,3,’’,‘1479827529.302’,‘1479827528.301’,’’,’’,’’)]
> 0x7fb5f401e000 – Probation passed - setting RTP source address to
– <SIP/002-0000012d> Playing ‘pls-try-call-later.gsm’ (language ‘en’)
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on ‘SIP/002-0000012d’ in macro ‘outisbusy’
== Spawn extension (from-internal, 7447874400728, 6) exited non-zero on ‘SIP/002-0000012d’
– Executing [h@from-internal:1] Hangup(“SIP/002-0000012d”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/002-0000012d’
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘HANGUP’,{ts ‘2016-11-22 09:12:52.644261’},’’,‘002’,‘002’,’’,‘7447874400728’,‘h’,‘from-internal’,‘SIP/002-0000012d’,’’,’’,3,’’,‘1479827528.301’,‘1479827528.301’,’’,’’,’’)]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_END’,{ts ‘2016-11-22 09:12:52.773489’},’’,‘002’,‘002’,’’,‘7447874400728’,‘h’,‘from-internal’,‘SIP/002-0000012d’,’’,’’,3,’’,‘1479827528.301’,‘1479827528.301’,’’,’’,’’)]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘LINKEDID_END’,{ts ‘2016-11-22 09:12:52.773503’},’’,‘002’,‘002’,’’,‘7447874400728’,‘h’,‘from-internal’,‘SIP/002-0000012d’,’’,’’,3,’’,‘1479827528.301’,‘1479827528.301’,’’,’’,’’)]

Please use insecure=invite
My understanding that you have a DID from that provider - right?
In this case you need to make a call to that DID somehow then we will need to analyze the incoming INVITE message you received from them.
Regular Asterisk logging is better to switch off for now.

yes, i am using did@serverip:5060

Generally you should provide SIP debugging, but 500 means they are admitting that their system is broken.

insecure=very is deprecated. The direct equivalent is insecure=invite,port. However, port is probably not needed, and invite only makes sense if you have a secret configured.

It would be very strange for an ITSP not to require a secret. With modern versions of Asterisk, it is clearer if you use remotesecret, rather than secret and insecure=invite.

NAT settings are generally determined by your end as much as the remote end.

callgroup and pickupgroup are purely local considerations.