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About the Asterisk SIP category
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2
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1040
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January 30, 2020
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PJSIP calling issues
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12
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90
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August 4, 2020
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Using established registration connection for incoming calls
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2
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20
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August 3, 2020
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PJSIP no audio on calls
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4
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50
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August 3, 2020
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Reading custom headers from REFER message
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5
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63
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August 2, 2020
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Sip not wotking in Asterisk server
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14
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198
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August 2, 2020
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Connect via SIP Trunk
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4
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56
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July 30, 2020
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Multiple peers from same servers are getting the same context
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12
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70
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July 28, 2020
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Asterisk 16 TLS - Bad file descriptor errors
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4
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97
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July 27, 2020
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SDP address set to loopback address when using trunk
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4
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117
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July 27, 2020
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Need help converting to pjsip
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9
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121
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July 27, 2020
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RTP Encryption, SDES, AES
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2
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41
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July 27, 2020
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Configuring chan_sip with sdes
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12
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143
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July 27, 2020
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In defense of chan_sip
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3
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108
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July 27, 2020
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RTPAUDIOQOS all zeros
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2
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54
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July 27, 2020
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Sip diversion to set Caller with Verizon
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8
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117
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July 26, 2020
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How To List RTP Address for Channel
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2
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44
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July 24, 2020
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FROM field in SIP packet header
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5
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59
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July 24, 2020
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Asterisk Realtime Database Creation
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7
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119
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July 24, 2020
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No dial tone, only one way listening
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4
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68
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July 24, 2020
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After receiving call no Sound coming at either End
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4
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44
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July 24, 2020
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(SIP/2.0 488 Not Acceptable Here), when configured with PJSIP/TLS
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4
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67
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July 24, 2020
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PJSIP syntax error exception when parsing 'From' header on line 4 col 54
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1
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34
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July 23, 2020
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Asterisk 16 - LTS - No more PJSIP-Support?
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3
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89
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July 23, 2020
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PJSIP Logger not working for the new commands
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6
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73
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July 22, 2020
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ContactStatus oscilating between Removed and Reachable
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8
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106
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July 22, 2020
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Voice Not received
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6
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60
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July 20, 2020
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DTMF cuts off audio
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1
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33
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July 20, 2020
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WARNING[1342][C-00000018]: chan_sip.c:24054 handle_response_invite: Received response: "Forbidden" from '"1001" <sip:1001@192.168.2.28>;tag=as11e7d521'
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5
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823
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July 20, 2020
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Sip set debug on
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2
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50
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July 17, 2020
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