thr3ads.net - asterisk users - [Asterisk-Users] Problems on making calls from one Gnophone to another through the local Asterisk Server [Oct 2003]

If this information is useful, please help other people find it:
Share via:

sheebaaggarwal@hfcl.com

2003-Oct-20 00:00 UTC

[Asterisk-Users] Problems on making calls from one Gnophone to another through the local Asterisk Server

Dear Members,

I am trying to make call from one Gnophone to another through the local
Asterisk Server.All the three systems have local IP Addresses
I created two users "sheeba" (extension 600) and "test"
(extension 602) in
iax.conf file:

[sheeba]
type=friend
auth=plaintext
host=dynamic
secret=sheeba
context=default
allow=gsm
permit=0.0.0.0/0.0.0.0
callerid="sheeba"<600>

[test]
type=friend
auth=plaintext
host=dynamic
secret=test
allow=gsm
context=default
permit=0.0.0.0/0.0.0.0
callerid="test"<602>

In the extension.conf file ,i made the following configuration:

exten => 600,1,Dial(IAX/sheeba/s|100|r)
exten => 600,2,Playback(demo-congrats)
exten => 600,102,Playback(demo-thanks)
exten => 600,3,Hangup

exten => 602,1,Dial(IAX/test/s|100|r)
exten => 602,2,Playback(demo-congrats)
exten => 602,102,Playback(demo-thanks)
exten => 602,3,Hangup

On making this configuration,i called user "test" with extension 602
from
user "sheeba" with extension 600.
I got the following log on server:

Accepting AUTHENTICATED call from 192.168.8.156, requested format = 2,
actual format = 2
    -- Executing Dial("IAX[sheeba@sheeba]/80",
"IAX/test/s|100|r") in new
stack
    -- Calling using options
'exten=s;callerid="sheeba"<600>;language=en;formats=2;capability=65283;version=1;adsicpe=0'
    -- Called test/s
    -- Call accepted by 192.168.8.62 (format GSM)
    -- Format for call is GSM
    -- IAX[test]/81 is ringing
    -- IAX[test]/81 answered IAX[sheeba@sheeba]/80
   -- Attempting native bridge of IAX[sheeba@sheeba]/80 and IAX[test]/81
    -- Channel 'IAX[sheeba@sheeba]/80' ready to transfer
    -- Channel 'IAX[test]/81' ready to transfer
    -- Releasing IAX[test]/81 and IAX[sheeba@sheeba]/80
    -- Hungup 'IAX[test]/81'
  == Spawn extension (default, 602, 1) exited non-zero on
'IAX[sheeba@sheeba]/80'
    -- Hungup 'IAX[sheeba@sheeba]/80'

In this log,it attempts the native bridge of "sheeba"with
"test",then it
gives both the channels are ready to transfer and then releases the
channels.The call is not getting through.What could be the possible error.
Pl help me asap by sending your valuable comments.This will be highly
appreciated.

Thanks and Regards,
Sheeba Aggarwal
Sr Engineer
HFCL R&D,Gurgaon,India

Sheeba Aggarwal

2003-Oct-20 20:51 UTC

head link

[Asterisk-Users] Problems on making calls from one Gnophone to another through the local Asterisk Server

Dear Members,

I am trying to make call from one Gnophone to another through the local
Asterisk Server.All the three systems have local IP Addresses
I created two users "sheeba" (extension 600) and "test"
(extension 602) in
iax.conf file:

[sheeba]
type=friend
auth=plaintext
host=dynamic
secret=sheeba
context=default
allow=gsm
permit=0.0.0.0/0.0.0.0
callerid="sheeba"<600>

[test]
type=friend
auth=plaintext
host=dynamic
secret=test
allow=gsm
context=default
permit=0.0.0.0/0.0.0.0
callerid="test"<602>

In the extension.conf file ,i made the following configuration:

exten => 600,1,Dial(IAX/sheeba/s|100|r)
exten => 600,2,Playback(demo-congrats)
exten => 600,102,Playback(demo-thanks)
exten => 600,3,Hangup

exten => 602,1,Dial(IAX/test/s|100|r)
exten => 602,2,Playback(demo-congrats)
exten => 602,102,Playback(demo-thanks)
exten => 602,3,Hangup

On making this configuration,i called user "test" with extension 602
from
user "sheeba" with extension 600.
I got the following log on server:

Accepting AUTHENTICATED call from 192.168.8.156, requested format = 2,
actual format = 2
    -- Executing Dial("IAX[sheeba@sheeba]/80",
"IAX/test/s|100|r") in new
stack
    -- Calling using options
'exten=s;callerid="sheeba"<600>;language=en;formats=2;capability=65283;version=1;adsicpe=0'
    -- Called test/s
    -- Call accepted by 192.168.8.62 (format GSM)
    -- Format for call is GSM
    -- IAX[test]/81 is ringing
    -- IAX[test]/81 answered IAX[sheeba@sheeba]/80
   -- Attempting native bridge of IAX[sheeba@sheeba]/80 and IAX[test]/81
    -- Channel 'IAX[sheeba@sheeba]/80' ready to transfer
    -- Channel 'IAX[test]/81' ready to transfer
    -- Releasing IAX[test]/81 and IAX[sheeba@sheeba]/80
    -- Hungup 'IAX[test]/81'
  == Spawn extension (default, 602, 1) exited non-zero on
'IAX[sheeba@sheeba]/80'
    -- Hungup 'IAX[sheeba@sheeba]/80'

In this log,it attempts the native bridge of "sheeba"with
"test",then it
gives both the channels are ready to transfer and then releases the
channels.The call is not getting through.What could be the possible error.
Pl help me asap by sending your valuable comments.This will be highly
appreciated.

Regards,

Sheeba Aggarwal
Senior Engineer,  HFCL R&D
286, Phase 2, Udyog Vihar
Gurgaon, Haryana
India -122016
 
Phone No: 91-124-2348846/842 - ext-122
email: sheebaaggarwal@hfcl.com



--------------------------------------------------------------------------------
 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20031020/a88a3634/attachment.htm

Sheeba Aggarwal

2003-Oct-20 20:51 UTC

head link

[Asterisk-Users] Problems on making calls from one Gnophone to another through the local Asterisk Server

Dear Members,

I am trying to make call from one Gnophone to another through the local
Asterisk Server.All the three systems have local IP Addresses
I created two users "sheeba" (extension 600) and "test"
(extension 602) in
iax.conf file:

[sheeba]
type=friend
auth=plaintext
host=dynamic
secret=sheeba
context=default
allow=gsm
permit=0.0.0.0/0.0.0.0
callerid="sheeba"<600>

[test]
type=friend
auth=plaintext
host=dynamic
secret=test
allow=gsm
context=default
permit=0.0.0.0/0.0.0.0
callerid="test"<602>

In the extension.conf file ,i made the following configuration:

exten => 600,1,Dial(IAX/sheeba/s|100|r)
exten => 600,2,Playback(demo-congrats)
exten => 600,102,Playback(demo-thanks)
exten => 600,3,Hangup

exten => 602,1,Dial(IAX/test/s|100|r)
exten => 602,2,Playback(demo-congrats)
exten => 602,102,Playback(demo-thanks)
exten => 602,3,Hangup

On making this configuration,i called user "test" with extension 602
from
user "sheeba" with extension 600.
I got the following log on server:

Accepting AUTHENTICATED call from 192.168.8.156, requested format = 2,
actual format = 2
    -- Executing Dial("IAX[sheeba@sheeba]/80",
"IAX/test/s|100|r") in new
stack
    -- Calling using options
'exten=s;callerid="sheeba"<600>;language=en;formats=2;capability=65283;version=1;adsicpe=0'
    -- Called test/s
    -- Call accepted by 192.168.8.62 (format GSM)
    -- Format for call is GSM
    -- IAX[test]/81 is ringing
    -- IAX[test]/81 answered IAX[sheeba@sheeba]/80
   -- Attempting native bridge of IAX[sheeba@sheeba]/80 and IAX[test]/81
    -- Channel 'IAX[sheeba@sheeba]/80' ready to transfer
    -- Channel 'IAX[test]/81' ready to transfer
    -- Releasing IAX[test]/81 and IAX[sheeba@sheeba]/80
    -- Hungup 'IAX[test]/81'
  == Spawn extension (default, 602, 1) exited non-zero on
'IAX[sheeba@sheeba]/80'
    -- Hungup 'IAX[sheeba@sheeba]/80'

In this log,it attempts the native bridge of "sheeba"with
"test",then it
gives both the channels are ready to transfer and then releases the
channels.The call is not getting through.What could be the possible error.
Pl help me asap by sending your valuable comments.This will be highly
appreciated.

Regards,

Sheeba Aggarwal
Senior Engineer,  HFCL R&D
286, Phase 2, Udyog Vihar
Gurgaon, Haryana
India -122016
 
Phone No: 91-124-2348846/842 - ext-122
email: sheebaaggarwal@hfcl.com



--------------------------------------------------------------------------------
 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20031020/1f45e0bd/attachment.htm

Reasonably Related Threads

Search for more reasonably related threads

asterisk users - Oct 2003 - Problems on making calls from one Gnophone to another through the local Asterisk Server

[Asterisk-Users] Problems on making calls from one Gnophone to another through the local Asterisk Server

[Asterisk-Users] Problems on making calls from one Gnophone to another through the local Asterisk Server

[Asterisk-Users] Problems on making calls from one Gnophone to another through the local Asterisk Server

Reasonably Related Threads