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lzwierko at gmail.com
Guest

Posted: Sun Jul 19, 2009 10:47 am Post subject: [Freeswitch-users] Dial up from confernece issue

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I want to use bgdial command to add a person to a already started conference (that is, call that person and when answered - add the channel to conference). The scenario is I have two sip clients registered in default context - 1000 and 1001. 1000 dials conference number (3001 in this case) and new conference is started. I want to dial out to second using bgdial, unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 Legged calls' message. Should I use the bgdial command differently? Or perhaps I should do this totally differently? Logs attached below. Thanks for any help, Lukasz freeswitch@Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000@192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear\|speak\|floor;0;0;300 freeswitch@Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001@192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001@192.168.0.1)] output: OK freeswitch@Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211@192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001@192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001@192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211@192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211@192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211@192.168.0.1 [CS_DESTROY] -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKYwMRAAoJED7LBosr0F2ullAH/j6AebaezM2/RQ4PVKeNbMEm yWYqC1bDmp5F56owRH6Vq7BRnXKB4roqV2NLFqLNRYwzq/S4bzc9p417/NckrACg DhmZ6tFd4ujLb6B1HvJMTKsDvnYCpn5EVCbENfKVIY4INDAcEYbncwUA21XxILI+ ztz+6qNPwOMOjY9aZaf1qpTcTcG2yn62mpvesmVeYS1vNpZFVpnQq4PrukDg+1xs N8EpJetP0FxhYzT/IiD9fS2wAzQSgJPgo0m7R4ezk/1NIF9f+o0irgc8zx+VgKw1 UhJ1FLhs8ObzhYclvwJxwTlG+ppI28uIVO8EItiPB4/ZhEjyPfNVHcqTvMG6wb4= =3Bm/ -----END PGP SIGNATURE----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

lzwierko at gmail.com
Guest

Posted: Sun Jul 19, 2009 3:26 pm Post subject: [Freeswitch-users] Dial up from confernece issue

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, sorry if you're getting this again, I'm not sure if this mail got deliverd to the mail-list (I didn't get a copy...) Anyway, I want to use bgdial command to add a person to a already started conference (that is, call that person and when answered - add the channel to conference). The scenario is I have two sip clients registered in default context - 1000 and 1001. 1000 dials conference number (3001 in this case) and new conference is started. I want to dial out to second using bgdial, unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 Legged calls' message. Should I use the bgdial command differently? Or perhaps I should do this totally differently? Logs attached below. Thanks for any help, Lukasz freeswitch@Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000@192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear\|speak\|floor;0;0;300 freeswitch@Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001@192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001@192.168.0.1)] output: OK freeswitch@Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001@192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211@192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001@192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001@192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001@192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211@192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211@192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211@192.168.0.1 [CS_DESTROY] -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKY3/FAAoJED7LBosr0F2uWo8H/iJEHblsAENCjRh/dsZvj9br Mq6txy7iafLE970XxvToaa0+FGBFxN+S6yQ6ampNPd8t+jl6WwC79Btwr+NLgXEc NcWpVQp65QxKxA+MgQOyqWIskcMcxdf4Uht3wuLPZtre0BpjcAFhykweYjOy1jFp AYAM61ShogHlpXtl9Z6upDvWPoOzdY4m13EM7f0NmpbC32Sg+OOULEtsxvSkZ8ah DBKDyDdXFo9iIcReDqjsu/kzAgrBsAZvOiEbSPoQTjZgzX+UrbgqIc+rhmP60vyt 8u8ufDgzh7MC/VQObHKHLe8e/Zbpaf+3JiGxZBtFyoUFyP3DjHoR5TYu5IsENPU= =QlRU -----END PGP SIGNATURE----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

brian at freeswitch.org
Guest

Posted: Sun Jul 19, 2009 3:29 pm Post subject: [Freeswitch-users] Dial up from confernece issue

If a single leg call gets a 302 you can't really "transfer" it
anywhere... What SVN rev are you on?

/b

On Jul 19, 2009, at 3:19 PM, Åukasz Zwierko wrote:

Quote:

Hi,

sorry if you're getting this again, I'm not sure if this mail got
deliverd to the mail-list (I didn't get a copy...)

Anyway,

I want to use bgdial command to add a person to a already started
conference (that is, call that person and when answered - add the
channel to conference).

The scenario is I have two sip clients registered in default context -
1000 and 1001. 1000 dials conference number (3001 in this case) and
new
conference is started. I want to dial out to second using bgdial,
unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1
Legged calls' message.

Should I use the bgdial command differently? Or perhaps I should do
this
totally differently? Logs attached below.

Thanks for any help,

Lukasz

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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lzwierko at gmail.com
Guest

Posted: Tue Jul 21, 2009 2:43 pm Post subject: [Freeswitch-users] Dial up from confernece issue

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Brian,

I've just updated to 14310 and it's the same. The thing seems that sofia
module rejects the call in quite early stage, so there is no 302 answer
from remote SIP peer (as no INVITE was sent).
Again, I'm exercising a very simple scenario with default FS
configuration (just downloaded from svn), so I don't really know what's
wrong here...
Perhaps there is a different way to attach a call to a existing conference?
Perhaps I should just originate new call (with the 'originate' command),
and when received, pass it it conference application with the
conference-id of the conference that I want to attach it to? Does that
make any sense?

Thanks

Å

freeswitch@Zwierko-laptop> conference list
API CALL [conference(list)] output:
Conference 3001-192.168.0.1 (1 member)
3;sofia/internal/1000@192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300

freeswitch@Zwierko-laptop> conference 3001-192.168.0.1 bgdial
sofia/default/1001@192.168.0.1
API CALL [conference(3001-192.168.0.1 bgdial
sofia/default/1001@192.168.0.1)] output:
OK

freeswitch@Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/1001@192.168.0.1
[b9fada7f-9c1d-4949-af8a-a8220ce
f9c5b]
2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/48228882211@192.168.0.1
[4f6b26dd-a0cb-2846-ad17-5f517e60e2e7]
2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing
TelkaSwitch->1001 in context public
2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1
Legged calls
2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup
sofia/internal/1001@192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create
outgoing channel, cause: NO_USER_RESPONSE
2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9
(sofia/internal/1001@192.168.0.1) Ended
2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close
Channel sofia/internal/1001@192.168.0.1 [CS_DESTROY]
2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/internal/48228882211@192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING]
2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session
10 (sofia/internal/48228882211@192.168.0.1) Ended
2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close
Channel sofia/internal/48228882211@192.168.0.1 [CS_DESTROY]

Brian West wrote:

Quote:

If a single leg call gets a 302 you can't really "transfer" it
anywhere... What SVN rev are you on?

/b

On Jul 19, 2009, at 3:19 PM, Åukasz Zwierko wrote:

Quote:

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Version: GnuPG v1.4.9 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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brian at freeswitch.org
Guest

Posted: Tue Jul 21, 2009 2:49 pm Post subject: [Freeswitch-users] Dial up from confernece issue

Its a 302 on a single leg call right?

/b

On Jul 21, 2009, at 2:38 PM, Åukasz Zwierko wrote:

Quote:

lzwierko at gmail.com
Guest

Posted: Wed Jul 22, 2009 12:52 am Post subject: [Freeswitch-users] Dial up from confernece issue

I'm not sure how this exactly works, but I suppose that it is a single
leg call, which upon answer would be attached to the conference (?)
somehow. But again, this call does not originate outside FS so what
would be the cause for 302?

2009/7/21 Brian West <brian@freeswitch.org>:

Quote:

Its a 302 on a single leg call right?

/b

On Jul 21, 2009, at 2:38 PM, £ukasz Zwierko wrote:

Quote:

brian at freeswitch.org
Guest

Posted: Wed Jul 22, 2009 5:58 am Post subject: [Freeswitch-users] Dial up from confernece issue

The far end you're calling is sending a 302 can you check the sip
traffic please.

/b

On Jul 22, 2009, at 12:45 AM, £ukasz Zwierko wrote:

Quote:

lzwierko at gmail.com
Guest

Posted: Fri Jul 24, 2009 2:22 am Post subject: [Freeswitch-users] Dial up from confernece issue

Ok Brian, you were right after all - I've had my X-lite incorrectly
configured, sorry for wasting your time.

thanks,

LZ

W dniu 22 lipca 2009 12:50 u¿ytkownik Brian West <brian@freeswitch.org> napisa³:

Quote:

The far end you're calling is sending a 302 can you check the sip
traffic please.

/b

On Jul 22, 2009, at 12:45 AM, £ukasz Zwierko wrote:

Quote:

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