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About the Asterisk SIP category
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1
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1726
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January 30, 2020
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SIP Trunk Between Asterisk and IP Office
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15
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31
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February 21, 2025
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Ssl connection with peer
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0
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10
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February 21, 2025
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Connection Trouble PJSIP Public IP
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0
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9
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February 21, 2025
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Help with QoS values
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2
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30
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February 20, 2025
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No response from asterisk
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10
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21
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February 20, 2025
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Problem registering speakers behind nat (sip)
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0
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8
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February 20, 2025
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Connecting two asterisk servers username mismatch
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6
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35
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February 19, 2025
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Call Drop in TLS transport
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3
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24
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February 19, 2025
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Issue: asterisk-pjsip over webrtc
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9
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33
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February 19, 2025
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Configure RTT - Asterisk 18
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0
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25
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February 18, 2025
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Webrtc call 404 Not Found
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10
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26
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February 18, 2025
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Two geteways FreePBX
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8
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26
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February 17, 2025
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Adding Custom Header to 183 Session Progress in Asterisk
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14
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63
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February 17, 2025
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How to enable video in Asterisk 22
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0
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17
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February 15, 2025
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Configuring trunk to make external call
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1
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17
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February 15, 2025
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Clients connect, but there is no sound on calls
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11
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54
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February 14, 2025
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Call is not getting routed to asterisk
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1
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12
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February 14, 2025
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Trying to send more than one outbound call via AMI, PJSIP only dials one
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1
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9
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February 14, 2025
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No such command pjsip show endpoints
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6
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35
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February 13, 2025
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Migrating to PJSIP - endpoint was not found
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4
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19
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February 13, 2025
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Pjsip, a solution to avoid this problem?
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17
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55
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February 11, 2025
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Asterisk SIP TRUNK DID
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1
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26
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February 11, 2025
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PJSIP channelstats command truncates 20-digit SIP numbers to 18 digits
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4
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26
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February 10, 2025
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Auto Answer SIP Calls with Asterisk on Ubuntu 20.04
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1
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15
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February 10, 2025
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No RTP Stream in PJSIP Calls (Asterisk in Docker on Windows)
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7
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45
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February 7, 2025
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Hold with SIP via TLS faulty
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0
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12
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February 6, 2025
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Opensips as mid registrar and asterisk as main registarar
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3
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21
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February 6, 2025
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STIR/SHAKEN doesn't continue for Anonymous calls
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6
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34
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February 5, 2025
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Audio cutting off after ring
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3
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20
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February 5, 2025
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