Difference between revisions of "How to connect to Huawei devices"
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Our main Asterisk <---> Patched Asterisk <---> Huawei softswitch | Our main Asterisk <---> Patched Asterisk <---> Huawei softswitch | ||
In '''/usr/src/asterisk/channels/chan_sip.c''' we commented lines | |||
if (!p->owner || !ast_internal_timing_enabled(p->owner)) | |||
ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n") | |||
as it is described in this [http://bugs.digium.com/view.php?id=6669 patch]. | |||
Revision as of 15:02, 25 January 2008
Huawei develops various devices which has nasty bugs in SIP implementation.
Huawei sends pagtches to Asterisk to cripple Asterisk to support their non-standard SIP implementation but do not fix their own implementations.
One we encountered is then in SIP INVITE message Asterisk sends line:
a=silenceSupp:off - - - - -
Huawei softswitch does not like it and returns error.
Related links:
- http://lists.digium.com/pipermail/asterisk-dev/2006-March/019067.html
- http://lists.digium.com/pipermail/asterisk-dev/2006-March/019156.html
- http://bugs.digium.com/view.php?id=6669
Provider which uses Huawei insisted we should patch (criple) Asterisk to make to work with their softdwitch. Instead we put middle server between our Asterisk server and their softswitch:
Our main Asterisk <---> Patched Asterisk <---> Huawei softswitch
In /usr/src/asterisk/channels/chan_sip.c we commented lines
if (!p->owner || !ast_internal_timing_enabled(p->owner))
ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n")
as it is described in this patch.
