Issues
- 0
Event Accept_incoming_call_completed should be sent after 183 (early media call)
#100 opened by virgilio-a-cunha - 0
- 0
Memory leak in ASR transcoding scenarios
#98 opened by virgilio-a-cunha - 0
- 0
ASR resource must support slinear codec
#92 opened by virgilio-a-cunha - 0
200OK with 2 contact headers
#93 opened by fvieira1977 - 0
- 0
- 0
200 OK for INVITE has no Contact header in Incoming Call scenario with early media
#89 opened by virgilio-a-cunha - 0
Play to conference scenario has a race condition that causes a segfault
#91 opened by virgilio-a-cunha - 0
Asterisk responds that the conference creation was successful even when conf_bridge was not created yet
#87 opened by virgilio-a-cunha - 0
SIP BYE content-length is not correct
#88 opened by virgilio-a-cunha - 0
RTP has wrong format when sent to ASR server
#85 opened by virgilio-a-cunha - 0
Outbound calls privacy
#86 opened by fvieira1977 - 0
User and bridge profiles defined in the configuration file are not applied in confbridge
#84 opened by virgilio-a-cunha - 0
- 0
- 0
STRICT RTP switch and block to one source and does not allow switch to another
#81 opened by fvieira1977 - 0
- 0
Music on hold provided from two different sources to same RTP destination port it is mixed
#79 opened by virgilio-a-cunha - 0
- 0
- 0
- 0
Sip usereqphone=yes does not add user=phone in From and P-Asserted-Identity headers
#75 opened by virgilio-a-cunha - 0
Multiple Byes sent after Refer
#74 opened by nuno-f-novo-alb - 0
transfer sip call issues Error 110
#72 opened by nuno-f-novo-alb - 0
Transfer call via SIP REFER with extra data
#70 opened by asapage - 1
- 0
- 0
- 1
Core dump on FAX reception with few hundred pages
#68 opened by asapage - 0
- 1
Core dump in load test with bridged channels
#65 opened by asapage - 0
- 0
MixMonitor does not load in legacy environments
#60 opened by asapage - 0
- 0
- 0
- 0
Channel leak after audio playback in ConfBridge
#59 opened by asapage - 0
- 0
- 0
- 0
SIP INFO request is not sent via SendText App if MESSAGE is not one of the allowed methods
#54 opened by asapage - 0
SIP headers are limited to 255 characters
#56 opened by asapage - 3
reading SIP-I over TCP
#53 opened by usmanbaiga - 0
- 2
no early media with Session Progress
#48 opened by usmanbaiga - 0
- 0
Rename the Patch52 folder and filename
#52 opened by asapage - 0