Hi,

I'm running freswitch version 1.4.4 and getting a strange codec error

when i call from a phone running CSIPSIMPLE(registered as user 1000) to 1005, the call hangs up and I get:

2014-06-05 08:31:59.168784 [INFO] switch_ivr_originate.c:1191 Sending early media

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[G722:9:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[PCMU:0:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[PCMA:8:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[GSM:3:8000:20:13200]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[G722:9:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[PCMU:0:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[PCMA:8:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[GSM:3:8000:20:13200]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[G722:9:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[PCMU:0:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[PCMA:8:8000:20:64000]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[GSM:3:8000:20:13200]

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3309 Set telephone-event payload to 101

2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3626 Set 2833 dtmf send/recv payload to 101

2014-06-05 08:31:59.168784 [ERR] mod_sofia.c:2201 CODEC NEGOTIATION ERROR.  SDP:

v=0

o=- 3610945918 3610945918 IN IP4 10.0.1.46

s=pjmedia

c=IN IP4 10.0.1.46

t=0 0

m=audio 4020 RTP/AVP 99 0 8 101

c=IN IP4 10.0.1.46

a=rtpmap:99 SILK/24000

a=fmtp:99 useinbandfec=0

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtcp:4021 IN IP4 10.0.1.46

2014-06-05 08:31:59.168784 [DEBUG] switch_core_session.c:907 Send signal sofia/internal/1000@10.0.1.233 [BREAK]

2014-06-05 08:31:59.168784 [NOTICE] switch_channel.c:3432 Hangup sofia/internal/1000@10.0.1.233 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

when i call using cmd 'originate user/1005 &bridge(user/1000)' The call goes through and I get:

2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3437 Audio Codec Compare [PCMU:0:8000:20:64000] ++++ is saved as a match

2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]

2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]

2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200]

2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3309 Set telephone-event payload to 101

2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:2343 Set Codec sofia/internal/sip:1000@10.0.1.46:51113 PCMU/8000 20 ms 160 samples 64000 bits

2014-06-05 08:25:18.073562 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1000@10.0.1.46:51113 Original read codec set to PCMU:0

2014-06-05 08:25:18.093570 [DEBUG] switch_core_media.c:3617 Set 2833 dtmf send payload to 101

2014-06-05 08:25:18.093570 [DEBUG] switch_core_media.c:4841 AUDIO RTP [sofia/internal/sip:1000@10.0.1.46:51113] 10.0.1.233 port 32636 -> 10.0.1.46 port 4016 codec: 0 ms: 20

2014-06-05 08:25:18.093570 [DEBUG] switch_rtp.c:3349 Starting timer [soft] 160 bytes per 20ms

2014-06-05 08:25:18.113587 [INFO] switch_core_media.c:5053 Activating RTCP PORT 4017

Even stranger, when I call using the same configuration under 1.2.17, the call goes through:

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[G722:9:8000:20:64000]

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[PCMU:0:8000:20:64000]

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[PCMA:8:8000:20:64000]

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[GSM:3:8000:20:13200]

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:3186 Set Codec sofia/internal/1000@10.0.1.233 PCMU/8000 20 ms 160 samples 64000 bits

2014-06-05 08:43:39.400774 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@10.0.1.233 Original read codec set to PCMU:0

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5446 Set 2833 dtmf send/recv payload to 101

2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:3445 AUDIO RTP [sofia/internal/1000@10.0.1.233] 10.0.1.233 port 23172 -> 10.0.1.46 port 4004 codec: 0 ms: 20

what could possibly be the reason for that? I have little understanding of codecs handling.

Thanks,

Shlomi Agiv

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