No Dialout from SNOM 320 on UCX 1

You are hitting the check key right, not trying to be funny but it does happen with some of our customers.


Avaya/Nortel/NEC/Asterisk/Access Control/CCTV/DSX/Acti

No problem. Yes I'm pressing the check key. Whenever attempting a call I get a message in the display that reads( Not available here) Disconnected. I'm wondering if the Snoms require a different dialing pattern in the outbound trunk!!!

Voice Connect Plus, Inc.
have a great day and a better tomorrow

The dial patterns are common for all phones (SIP or other) - there are no special patterns for SNOM phones.

You may want to check if the dial request actually makes it to the UCx system (go to PBX - PBX Configuraton - Log file to view the current tail of the log, then try to make the call and redisplay the tail to see if any entries were generated based on the SNOM request).

If the request does not leave the phone, your problem is in the phone's configuration. I suspect that will be the case.

If the request reaches the UCx system, you should see a line with the text Dial(<identification of channel for your phone>,<dialed destination>) close to the bottom of the file. Check the destination digits - they should match your outbound dial pattern.

Below is the log I'm getting whenever I attempt a call. Have no clue whatsoever as to what it means.



[Feb 8 18:14:02] VERBOSE[3138] netsock2.c: == Using SIP RTP TOS bits 184
[Feb 8 18:14:02] VERBOSE[3138] netsock2.c: == Using SIP RTP CoS mark 5
[Feb 8 18:14:02] WARNING[3138] chan_sip.c: Rejecting audio media offer due to invalid or unsupported syntax: audio 49454 TCP/RTP/AVP 9 0 8 3 9

Voice Connect Plus, Inc.
have a great day and a better tomorrow

Based on the TCP/RTP/AVP, it looks like you have the phone configured to use RTP/AVP stream that uses the framing over TCP. RTP is normally sent over UDP. The media offer is rejected - clearly, this framing method is not supported by UCx. You should change the configuration of your phone to use normal RTP over UDP.

Hey UCXGUY, after making the RTP over UDP change I continued to get the following error message.
[Feb 8 19:29:45] VERBOSE[3138] netsock2.c: == Using SIP RTP TOS bits 184
[Feb 8 19:29:45] VERBOSE[3138] netsock2.c: == Using SIP RTP CoS mark 5
[Feb 8 19:29:45] WARNING[3138] chan_sip.c: We are requesting SRTP for audio, but they responded without it!
[Feb 8 19:30:01] VERBOSE[2984] asterisk.c: -- Remote UNIX connection
[Feb 8 19:30:01] VERBOSE[10526] asterisk.c: -- Remote UNIX connection disconnected
After going through my SIP settings I noticed this what set to off [Force sendrecv on INVITE with no SDP:] I changed the setting to on and all is well.Thanks so much for the help brother. I can guarantee you I'll never revisit this again. LOL Have a great weekend!!!!!

Voice Connect Plus, Inc.
have a great day and a better tomorrow

Hi atdjr123
I am having the same problem with a snom 360 set.
Where do you change the force send rcv setting ?
Thanks

Hey Engenius, you have to access the snom via web browser. Enter the IP address of the snom. Once you have access click on the identity of your snom ( indentity 1-12 ). Once you are in the indentity configuration click on SIP and scroll down to to ( Force sendrcv on Invite with no SDP ). Change to on then scroll down to apply and then save. Hope this helps you brother!!!

Voice Connect Plus, Inc.
have a great day and a better tomorrow