[asterisk-users] strange things on call transfer
MIVOC Systemes SAS
mivoc at wanadoo.fr
Wed Mar 14 03:02:40 MST 2007
Hi Chris,
I have the same problem because the pound(#) transfer with a Matra PABX.
I search a solution to replace the # touch that cause many problems
In France, we use the R touch to call transfer ...
In any way , I'm interest in your setup, your problems and your solutions
because we have a same project
( Different PABX Alacatel and Matracom !! )
( I have a ASUS-ISDN Card in the Asterisk Box Centos 4.4 )
Minh VO
MIVOC Systemes
----- Original Message -----
From: "Christoph Fürstaller" <christoph.fuerstaller at kurtkrenn.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Wednesday, March 14, 2007 10:44 AM
Subject: Re: [asterisk-users] strange things on call transfer
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi,
>
> I think I solved the 'problem'. The problem is my phone, Thomson
> ST2030S. I set the DTMF Mode to RFC2833, as I usually do. On the Thomson
> phone, I get this error when I try to hit/send a number or #. If I set
> DTMF Mode to SIP INFO, it works. the interesting thing is, when I call
> voicemailbox (with DTMF Mode RFC2833) I can login to the
> voicemailsystem, but if I hit # during a call, to transfer it, I get
> warnings and the transfer doesn't work. Seemes like this is a bug or bad
> implementation in the phone?
>
> My setup:
> Asterisk (NT) is connected via 1xE1 (Sangoma A104) to Alcatel 4400 PBX
> (TE). Asterisk gets clock from Alcatel. Do you need that to provide help
> or are you interested how I did it?
>
> chris...
>
> MIVOC Systemes SAS schrieb:
> > Hello,
> >
> > Can you tell about your configuration ( connection between Alacaltel
4400
> > and Asterisk ?) : what hardware ? what config files ?
> >
> > Thank you
> >
> > Minh VO
> > MIVOC Systemes SAS
> > FRANCE
> > ----- Original Message -----
> > From: "Christoph Fürstaller" <christoph.fuerstaller at kurtkrenn.com>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Wednesday, March 14, 2007 9:16 AM
> > Subject: [asterisk-users] strange things on call transfer
> >
> >
> > 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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> Aucun virus connu a ce jour par nos services n'a ete detecte.
> >>
> >>
>
>
> --------------------------------------------------------------------------
--
> > ----
>
>
> <-- 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'
> >>
> >>
>
>
>
> --------------------------------------------------------------------------
--
> > ----
>
>
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> >>
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>
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>
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>
> - --
> Dipl.-Ing. Kurt Krenn - IT-Beratung
> Franz-Josef-Strasse 33/4/43, 5020 Salzburg
> Tel: +43 662 879512 Fax: +43 662 875960
> IP-Tel: +43 780 kkrenn (557366)
> Email: c.fuerstaller at kurtkrenn.com
> sip: c.fuerstaller at kurtkrenn.com
>
> -----BEGIN PGP SIGNATURE-----
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> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFF98PfR0exH8dhr/YRAqvHAJ9gUIcu7pbSIUxLZ9cxxKvH24FHrwCgq3nT
> wH1Tb8H7+1/rFXCwowy2boI=
> =vtcf
> -----END PGP SIGNATURE-----
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> --------------------------------------------------------------------------
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> Orange vous informe que cet e-mail a ete controle par l'anti-virus mail.
> Aucun virus connu a ce jour par nos services n'a ete detecte.
>
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