search for: ast_rtp_new_source

Displaying 3 results from an estimated 3 matches for "ast_rtp_new_source".

2008 Jul 17
1
OpenH323 and ptlib version for asterisk 1.4.21.1
Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser
2011 Jan 12
1
DTMF not being heard correctly by far end conference system
...write format alaw [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 1713844722 to 565606422 due to a source change [Jan 12 23:13:55] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF begin on channel (IAX2/419-13088) [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update [Jan 12 23:13:55] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639 [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 565606422 to 226872656 due to a...
2010 Aug 23
2
Make a transfer for external line.
...ure_interpret: Feature interpret: chan=SIP/gustavo-00000002, peer=DAHDI/65-1, code=#1, sense=2, features=2, dynamic=# [Aug 18 09:16:58] DEBUG[4756]: features.c:1948 feature_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=#1 [Aug 18 09:16:58] DEBUG[4756]: rtp.c:2648 ast_rtp_new_source: Setting the marker bit due to a source update -- Started music on hold, class 'default', on channel 'SIP/gustavo-00000002' [Aug 18 09:16:58] DEBUG[4756]: channel.c:3710 set_format: Set channel DAHDI/65-1 to write format ulaw -- <DAHDI/65-1> Playing 'pbx-trans...