[asterisk-users] strange things on call transfer

Wed Mar 14 01:16:36 MST 2007

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Hi,

I'm setting up an Asterisk system which is connected to an Alcatel 4400
PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
call by hitting the # key, I get this messages and nothing happens on
the phone:

WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame
that isn't a multiple of 50 bytes long from RTP (4)?

But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I
tet this error and the call is hung up, when transfering:

Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
find a codec translation path from ilbc to slin
Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
find a codec translation path from ilbc to slin
Mar 14 09:04:58 WARNING[30436]: channel.c:2752
ast_channel_make_compatible: No path to translate from Zap/31-1(72) to
SIP/374-08199e50(1024)
Mar 14 09:04:58 WARNING[30436]: channel.c:3632 ast_channel_bridge: Can't
make Zap/31-1 and SIP/374-08199e50 compatible
Mar 14 09:04:58 WARNING[30436]: res_features.c:1385 ast_bridge_call:
Bridge failed on channels Zap/31-1 and SIP/374-08199e50

Attached there is a SIP debug from such a call (with ilbc loaded)

I hope someone can help me.

chris...
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-------------- next part --------------
<-- SIP read from 172.28.20.4:2051:
INVITE sip:374 at 172.28.2.30;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom300/6.2.3
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 1775117380 1775117380 IN IP4 172.28.20.4
s=call
c=IN IP4 172.28.20.4
t=0 0
m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

--- (18 headers 19 lines) ---
Using INVITE request as basis request - 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
Sending to 172.28.20.4 : 2051 (NAT)
Reliably Transmitting (no NAT) to 172.28.20.4:2051:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport;received=172.28.20.4
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>;tag=as49d79f99
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xx", nonce="0986769d"
Content-Length: 0

---
Scheduling destruction of call '3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7' in 15000 ms
Found user '104'
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
ACK sip:374 at 172.28.2.30;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>;tag=as49d79f99
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>;flow-id=1
Content-Length: 0

--- (9 headers 0 lines) ---
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
INVITE sip:374 at 172.28.2.30;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom300/6.2.3
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="104",realm="xxx.xx",nonce="0986769d",uri="sip:374 at 172.28.2.30;user=phone",response="0e5cb3b8871eff4c6670a6f277ef3594",algorithm=md5
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 1775117380 1775117380 IN IP4 172.28.20.4
s=call
c=IN IP4 172.28.20.4
t=0 0
m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

--- (19 headers 19 lines) ---
Using INVITE request as basis request - 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
Sending to 172.28.20.4 : 2051 (NAT)
Found user '104'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 172.28.20.4:59502
Peer video RTP is at port 172.28.20.4:65535
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x102 (gsm|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 374 in intern (domain 172.28.2.30;user=phone)
list_route: hop: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>
Transmitting (no NAT) to 172.28.20.4:2051:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:374 at 172.28.2.30>
Content-Length: 0

Transmitting (no NAT) to 172.28.20.4:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>;tag=as31ca24ed
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:374 at 172.28.2.30>
Content-Length: 0

---
    -- SIP/374-081b0378 answered SIP/104-0819a1f8
We're at 172.28.2.30 port 16206
Video is at 172.28.2.30 port 18568
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.28.20.4:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>;tag=as31ca24ed
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:374 at 172.28.2.30>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 29900 29900 IN IP4 172.28.2.30
s=session
c=IN IP4 172.28.2.30
t=0 0
m=audio 16206 RTP/AVP 18 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
ACK sip:374 at 172.28.2.30 SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-hjacxzcexdgk;rport
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>;tag=as31ca24ed
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>;flow-id=1
Content-Length: 0

--- (9 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 172.28.20.4:2051:
OPTIONS sip:104 at 172.28.20.4:2051;line=7p0ffzkf SIP/2.0
Via: SIP/2.0/UDP 172.28.2.30:5060;branch=z9hG4bK7e2b5210;rport
From: "asterisk" <sip:asterisk at xxx.xx>;tag=as68d9b69b
To: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>
Contact: <sip:asterisk at 172.28.2.30>
Call-ID: 30e124937558a11d3e9f397947b378b8 at xxx.xx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 Mar 2007 07:45:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.2.30:5060;branch=z9hG4bK7e2b5210;rport=5060
From: "asterisk" <sip:asterisk at xxx.xx>;tag=as68d9b69b
To: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>
Call-ID: 30e124937558a11d3e9f397947b378b8 at xxx.xx
CSeq: 102 OPTIONS
Contact: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>;flow-id=1
User-Agent: snom300/6.2.3
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Length: 0

--- (14 headers 0 lines) ---
Destroying call '30e124937558a11d3e9f397947b378b8 at xxx.xx'
Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?

<-- SIP read from 172.28.20.4:2051:
BYE sip:374 at 172.28.2.30 SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-e9vrp3jilhg7;rport
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>;tag=as31ca24ed
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:104 at 172.28.20.4:2051;line=7p0ffzkf>;flow-id=1
User-Agent: snom300/6.2.3
RTP-RxStat: Total_Rx_Pkts=350,Rx_Pkts=350,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=355,Tx_Pkts=355,Remote_Tx_Pkts=0
Content-Length: 0

--- (12 headers 0 lines) ---
Sending to 172.28.20.4 : 2051 (NAT)
Transmitting (NAT) to 172.28.20.4:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-e9vrp3jilhg7;received=172.28.20.4;rport=2051
From: "Test User3" <sip:104 at 172.28.2.30>;tag=48rd3g03e1
To: <sip:374 at 172.28.2.30;user=phone>;tag=as31ca24ed
Call-ID: 3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:374 at 172.28.2.30>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

---
  == Spawn extension (intern, h, 109) exited non-zero on 'SIP/104-0819a1f8'
Destroying call '3c30f7fb186a-n2ptt0oiuivb at snom300-0004132520F7'