Hello All,
Any one knows about what version of Asterisk that support to implement the ToIP ( Text Over IP ) ? Currently i am using asterisk 1.4.12 but always getting error. The errror says that there is supported SDP to send text.
All any kind of answer will be appreciate,
Best Regards
Tpx99
Howdy,
Support for T.140 was added in Asterisk 1.6.0.
Currently, Asterisk 1.6.0 is marked for EOL on October 1 of this year. We encourage you, if you want to use something from the 1.6 series, to use 1.6.2 as it will not go EOL until December of 2011.
The 1.8 release of Asterisk, marked as Long Term Support, will be supported for 5 years after release - currently it is in beta.
So, we would encourage you to either user 1.6.2.x or 1.8.x release of Asterisk if you need T.140 support.
Cheers.
Hello Malcom,
Very well thanks for your comment, Okay i’ll try the asterisk 1.6.2.x
Look at for the log below as i use asterisk 1.4.12 version to implement ToIP
asterisk*CLI>
– Registered SIP ‘2000’ at 192.168.1.4 port 5060 expires 3600
– Saved useragent “SIPcon1” for peer 2000
– Registered SIP ‘2001’ at 192.168.1.1 port 5060 expires 3600
– Saved useragent “SIPcon1” for peer 2001
[Aug 14 15:51:24] WARNING[2986]: chan_sip.c:5159 process_sdp: Unsupported SDP media type in offer: text 1026 RTP/AVP 99 98
– Executing [2001@telpun:1] Dial(“SIP/2000-09596808”, “SIP/2001”) in new stack
– Called 2001
– SIP/2001-0959a780 is ringing
Really destroying SIP dialog ‘499f72fec54e2e6240ef337928007c58@192.168.1.4’ Method: REGISTER
– SIP/2001-0959a780 answered SIP/2000-09596808
– Native bridging SIP/2000-09596808 and SIP/2001-0959a780
– Got SIP response 486 “Busy here” back from 192.168.1.1
== Spawn extension (telpun, 2001, 1) exited non-zero on ‘SIP/2000-09596808’
[Aug 14 15:51:35] ERROR[7379]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
Really destroying SIP dialog ‘0dec65ee1c7da361577bdeee398380fc@192.168.1.2’ Method: INVITE
– Got SIP response 486 “Busy here” back from 192.168.1.4
Really destroying SIP dialog ‘df406653c4cc0383c0d4331c79a81e9d@192.168.1.4’ Method: ACK
Really destroying SIP dialog ‘25e61d1d89805f3c72c9626b8ec2b0b2@192.168.1.1’ Method: REGISTER
From the log i think i used wrong version of asterisk as you encourage to try asterisk version 1.6.2.x or 1.8.x.x
Warm Regards,
Tpx99
Dear Malcolm and All,
Did you heve expereince or ever try to implement ToIP with SIPCon1 ?
Why i am still can not send a text to another Textphone ( SIPCon1 ) ? But i can send a voice very clear…
Regards,
tpx99