[asterisk-users] Question on ExternalMedia and the codec

2021-10-12 Thread Dan Cropp
We tell asterisk to use the slin format for ExternalMedia.  However, the 
unicast channel is selecting ulaw formatand the RTP data is indicating it's 
ulaw format.

Anyone know why ulaw format would be on chosen?


[10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is 
/ari/channels/externalMedia?app=a2519b4b-4d90-4d18-906b-717d02f8d569&external_host=192.168.32.148:8080&format=slin
[10/12 16:13:39.396] DEBUG[1665] http.c: match request 
[ari/channels/externalMedia] with handler [static] len 6
[10/12 16:13:39.396] DEBUG[1665] http.c: match request 
[ari/channels/externalMedia] with handler [ari] len 3
[10/12 16:13:39.396] DEBUG[1665] http.c: Match made with [ari]
[10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for 
channels/externalMedia
[10/12 16:13:39.396] DEBUG[1665] res_ari.c:   Finding handler for channels
[10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari deviceStates:  
Didn't match channels
[10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari applications:  
Didn't match channels
[10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari channels:  
Explicit match with channels
[10/12 16:13:39.396] DEBUG[1665] res_ari.c:   Finding handler for externalMedia
[10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels create:  
Didn't match externalMedia
[10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels 
channelId:  Matched wildcard.
[10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels 
externalMedia:  Explicit match with externalMedia
[10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our 
source address is '192.168.33.34'.
[10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: Using engine 'asterisk' for RTP 
instance '0x7fef60018320'
[10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) RTP 
allocated port 12226
[10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE 
creating session 192.168.33.34:12226 (12226)
[10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE create
[10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add 
system candidates
[10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add 
candidate: 192.168.33.34:12226, 2130706431
[10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: RTP instance '0x7fef60018320' is 
setup and ready to go
[10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:  : 
Formats: (none)
[10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:  Channel is being 
initialized or destroyed
[10/12 16:13:39.396] DEBUG[1665] stasis.c: Creating topic. name: 
channel:1634055219.4, detail:
[10/12 16:13:39.396] DEBUG[1665] stasis.c: Topic 'channel:1634055219.4': 
0x7fef6008d170 created
[10/12 16:13:39.396] DEBUG[1665] channel.c: Channel 0x7fef6008a910 
'UnicastRTP/192.168.32.148:8080-0x7fef60018320' allocated
[10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our 
source address is '192.168.33.34'.
[10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:  
UnicastRTP/192.168.32.148:8080-0x7fef60018320: Formats: (ulaw)
[10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:  New topology set
[10/12 16:13:39.396] DEBUG[1665] res_stasis.c: 
a2519b4b-4d90-4d18-906b-717d02f8d569: Subscribing to 1634055219.4
[10/12 16:13:39.396] DEBUG[1665] stasis/app.c: Channel '1634055219.4' is 1 
interested in a2519b4b-4d90-4d18-906b-717d02f8d569
[10/12 16:13:39.396] DEBUG[1665] http.c: HTTP keeping session open.  
status_code:200
[10/12 16:13:39.396] DEBUG[1666] stasis/app.c: Channel '1634055219.4' is 2 
interested in a2519b4b-4d90-4d18-906b-717d02f8d569

Have a good day!
Dan

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[asterisk-users] Asterisk 16 Realtime outbound register

2021-10-12 Thread Antony Stone
Hi.

I'm using Asterisk 16 with a MySQL realtime DB, containing both outbound 
registrations to other PBXs (Asterisk as SIP client) and inbound accounts for 
clients to register to (Asterisk as SIP server).

All in all, working well.

However, I just had a requirement to register outbound to a PBX on port 5070 
instead of 5060, and placing "5070" in the "port" field of the realtime 
database made no difference - Asterisk continued to try to register to port 
5060.

If I add ":5070" to the end of the "host" field, Asterisk correctly registers 
to port 5070 on the remote PBX.

So, what is the "port" field in the SIP peers DB table for, and how can I 
correctly specify the hostname and the port number separately?


Thanks,


Antony.

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