10-07-2013 02:47 PM - edited 03-19-2019 07:22 AM
Hello all...
My question\issue is 2 fold I guess...
I'm looking to find out why when I call my CUCM DN, enabled with Device Mobility (cellphone), it will not work when called from an extension from my FreePBX Asterisk box. It does not ring the configured cell phone as it should. This works fine when calling the same DN from another phone on the CUCM side, and, when I call in to my Asterisk system from my PSTN DID....which is weird why that works but calling from an "inside" Asterisk extension does not! The phones on each system are able to call eachother fine...I have a single SIP trunk set up between them. The Asterisk system acts like a gateway for PSTN calls to the outside world (SIP trunk provider)- the Cisco phones call out to PSTN fine also.
So..I fired up RTMT, and looked at a session trace. I think where it's stopping is the point where it WANTS to call the outside mobile- I get a 401 Unauthorized. See below.
2013/10/07 17:18:45.584|SIPT|21794532|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.14^192.168.0.3^*|26|b20ee00-25312535-2-500a8c0@192.168.0.5|401 Unauthorized
I was wondering also if there was another log or trace I can view\execute to get some more robust information as you do with Asterisk- like where it references the called and calling numbers, DID, etc..... that would be a further help to pin point what might be wrong.
Any help is appreciated--thanks in advance
Dennis
CUCM 8.6
Asterisk V 11.4
10-08-2013 12:42 AM
How is your CUCM connected to FreePBX ? Is it directly connected or you have an intermediate device sitting there. Can you specify what is the call flow. Also, if you can attach CUCM details traces with calling, called party and time of call, it would provide more details on what is happening.
10-08-2013 04:22 PM
have you tried to configure Asterisk to allow anonymous connections on your trunk between cucm and asterisk? CUCM cant deal with authentication and a CUBE would need to sit in the middle to take care of that.
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10-09-2013 04:20 PM
minkdennis...
I believe the trunk connecting the two systems is anonymous- this is my Peer and User on the Asterisk side
Peer
type=peer
transport=udp
qualify=yes
insecure=port
host=192.168.0.5
allow=ulaw
User
insecure=port
host=192.168.0.5
context=from-trunk
allow=ulaw
10-09-2013 04:16 PM
mkchandak....
CUCM is directly connected- sam LAN- same switch- nothing in between. Below is a call log for the call in question.
This is the flow basically
Free PBX extension- 1970- calls CUCM DN 3001- phone rings- I do not pick up- after 10 or so seconds the cell should ring- (Device Mobility)- it does not. I just let it ring some more and hangup.
See the Unauthorized after the second invite. Wish there was more info and clarification of what exactly is not authorized-- the asterisk extension? The trunk attempting to call the cell phone?? And like I said, it all works fine if I ring into Asterisk from my PSTN DID..... or from another CCUM registered phone. Any other logs I could be looking at? I'm fairly new to CUCM.
2013/10/09 17:41:52.117|SIPT|0|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.24^192.168.0.3^*|45|7f332e0a19c420b07dcb41ef3977c849@dtech.dnsalias.com|INVITE
2013/10/09 17:41:52.151|SIPT|0|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.24^192.168.0.3^*|46|7f332e0a19c420b07dcb41ef3977c849@dtech.dnsalias.com|100 Trying
2013/10/09 17:41:52.267|CC|SETUP|22487935|22487936|@dtech.dnsalias.com|3001|3001
2013/10/09 17:41:52.315|CC|OFFERED|22487935|22487936|1970|3001|3001|To_FreePBX|SEP0021A0857BE9
2013/10/09 17:41:52.320|SIPT|22487935|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.24^192.168.0.3^*|47|7f332e0a19c420b07dcb41ef3977c849@dtech.dnsalias.com|180 Ringing
2013/10/09 17:41:56.425|SIPT|22487936|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.24^192.168.0.3^*|48|9d0e2f80-2551cda4-1-500a8c0@192.168.0.5|INVITE
2013/10/09 17:41:56.428|SIPT|22487936|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.25^192.168.0.3^*|49|9d0e2f80-2551cda4-1-500a8c0@192.168.0.5|401 Unauthorized
2013/10/09 17:41:56.429|SIPT|22487936|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.25^192.168.0.3^*|50|9d0e2f80-2551cda4-1-500a8c0@192.168.0.5|ACK
2013/10/09 17:42:39.832|SIPT|22487935|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.26^192.168.0.3^*|51|7f332e0a19c420b07dcb41ef3977c849@dtech.dnsalias.com|CANCEL
2013/10/09 17:42:39.832|SIPT|22487935|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.26^192.168.0.3^*|52|7f332e0a19c420b07dcb41ef3977c849@dtech.dnsalias.com|200 OK
2013/10/09 17:42:39.837|SIPT|22487935|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.26^192.168.0.3^*|53|7f332e0a19c420b07dcb41ef3977c849@dtech.dnsalias.com|487 Request Cancelled
2013/10/09 17:42:39.839|CC|RELEASE|22487935|22487936|16
2013/10/09 17:42:39.841|SIPT|22487935|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.27^192.168.0.3^*|54|7f332e0a19c420b07dcb41ef3977c849@dtech.dnsalias.com|ACK
10-09-2013 06:23 PM
apparently in FreePBX there is a general setting "allow anonymous inbound calls" somewhere, but you might want to post that on a more approproate forum.
http://www.freepbx.org/forum/general-help/incoming-call-from-sip-trunk-failing
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11-05-2013 01:53 AM
Hi
It is only Cisco video phone(9971) is not able to make call through SIP trunk. Other Cisco phones can reach extension through SIP trunk. Still need to "allow anonymous inbound calls" on Asterisk ?
Rgds
Rajesh Kumar
01-20-2014 02:44 PM
Rajesh....picking this up again...I know it's been a while. I tried allowing anonymous inbound calls on my Asterisk server...no difference. The call out to the mobile is never made\allowed ... again, Device Mobility enabled.
This is trace of a sucessfull call from Freepbx to the CUCM extension in question (3001). The difference here is I answered the call...before it would have rang the mobile. I still see the 401 Unauthorized from from FreePBX though...but then CUCM ACKs it, and sends the 200 OK anyway. Just frustrating why it won't ring the mobile.
I feel like I'm missing something- I tried different CSSs etc...and there is no digest auth on the SIP to trunk from CUCM to FreePBX.... any help guys...would love to solve this thing just on principle! Dennis
2014/01/20 17:28:43.644|SIPT|0|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1737^192.168.0.3^*|3345|66b9646933c17aa76683396d0e23fb5d@dtech.dnsalias.com|INVITE
2014/01/20 17:28:43.645|SIPT|0|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1737^192.168.0.3^*|3346|66b9646933c17aa76683396d0e23fb5d@dtech.dnsalias.com|100 Trying
2014/01/20 17:28:43.647|CC|SETUP|24121072|24121073|@dtech.dnsalias.com|3001|3001
2014/01/20 17:28:43.650|CC|OFFERED|24121072|24121073|1970|3001|3001|To_FreePBX|SEP0021A0857BE9
2014/01/20 17:28:43.652|SIPT|24121072|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1737^192.168.0.3^*|3347|66b9646933c17aa76683396d0e23fb5d@dtech.dnsalias.com|180 Ringing
2014/01/20 17:28:47.667|SIPT|24121073|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1737^192.168.0.3^*|3348|3916ff00-2dd1a31f-119d-500a8c0@192.168.0.5|INVITE
2014/01/20 17:28:47.668|SIPT|24121073|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1738^192.168.0.3^*|3349|3916ff00-2dd1a31f-119d-500a8c0@192.168.0.5|401 Unauthorized
2014/01/20 17:28:47.669|SIPT|24121073|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1738^192.168.0.3^*|3350|3916ff00-2dd1a31f-119d-500a8c0@192.168.0.5|ACK
2014/01/20 17:28:48.631|SIPT|24121072|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,56,1.6273^192.168.0.42^SEP0021A0857BE9|3351|66b9646933c17aa76683396d0e23fb5d@dtech.dnsalias.com|200 OK
2014/01/20 17:28:48.633|SIPT|24121072|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1739^192.168.0.3^*|3352|66b9646933c17aa76683396d0e23fb5d@dtech.dnsalias.com|ACK
2014/01/20 17:28:55.415|SIPT|24121072|UDP|OUT|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,56,1.6276^192.168.0.42^SEP0021A0857BE9|3353|66b9646933c17aa76683396d0e23fb5d@dtech.dnsalias.com|BYE
2014/01/20 17:28:55.418|SIPT|24121072|UDP|IN|192.168.0.5|5060|To_FreePBX|192.168.0.3|5060|1,100,230,1.1740^192.168.0.3^*|3354|66b9646933c17aa76683396d0e23fb5d@dtech.dnsalias.com|200 OK
2014/01/20 17:28:55.418|CC|RELEASE|24121072|24121073|16
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