subject:"Re\: \[Asterisk\-Users\] a beginner's SIP question .."

Re: [Asterisk-Users] a beginner's SIP question ..

2003-06-04 Thread Michael Manousos

Hi,

Dave Alan Caruana wrote:
sorry i'm sending so many emails, I always think of something
exactly after i've pressed Send .. please be patient with me :)
 
I also have OH323 installed, supposedly correctly, and the same
gateway I want to connect to on SIP also supports H323, however
i do not know what the dial command line for H323 is .. i'm trying
 
exten = 1304,1,Dial(OH323/216.52.153.206) ;ring
but I actually want to dial extension 723 on the remote end,
First, make sure to specify a codec type, in oh323.conf, that is
supported by the gateway.
If a gatekeeper is used and the gateway and Asterisk are
registered on this gatekeeper, then you should do:
exten = 1304,1,Dial(OH323/723)

If there is no gatekeeper involved, do:

exten = 1304,1,Dial(OH323/[EMAIL PROTECTED])


so this is surely not right.. current messages i'm getting
from Asterisk are these :
 
*CLI dial 1304
-- Executing Dial(OSS/dsp, OH323/216.52.153.206) in new stack
*CLI   0:03.623   H323 Cleaner H323Connection 
ip$localhost/9771 terminated.
ERROR[1232188736]: File chan_oh323.c, Line 610 (oh323_call): H323:0: 
Could not call 216.52.153.206.
-- Couldn't call 216.52.153.206
-- Hungup 'H323:0'
  == Everyone is busy at this time
help *very* welcome ;)
 
cheers
Dave


Michael.


- Original Message -
*From:* Dan mailto:[EMAIL PROTECTED]
*To:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
*Sent:* Friday, May 30, 2003 7:50 PM
*Subject:* Re: [Asterisk-Users] a beginner's SIP question ..
Hi Dave,
 
If you have registered the SIP phone with Asterisk, then you must
have a line like:
 
exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207
mailto:SIP/[EMAIL PROTECTED],52,153.207)
 
in extensions.conf file
 
Then call 555 from the SIP phone to access the destination.
 
BR,
Dan

- Original Message -
*From:* Dave Alan Caruana mailto:[EMAIL PROTECTED]
*To:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
*Sent:* Friday, May 30, 2003 6:21 PM
*Subject:* Re: [Asterisk-Users] a beginner's SIP question ..
I have included a dump of the debug info ...
what I am trying to do is route a call from sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
If i dial direct from the sip phone to the gateway it works fine
.. so
I do not think there is any incompatibility there.
Calls don't go through though ...
 
please help!!!
 
cheers
Dave
 
 
*CLI -- Executing Dial(SIP/217.168.168.49:5060,
SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
-- SIP/216.52.153.207-eca2 answered SIP/217.168.168.49:5060
-- Attempting native bridge of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-eca2
WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt):
Maximum retries exceeded on call
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] for seqno 1 (Response)
  == Spawn extension (default, 1303, 1) exited non-zero on
'SIP/217.168.168.49:5060'
WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt):
Maximum retries exceeded on call
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] for seqno 1 (Response)
-- Executing Dial(SIP/217.168.168.49:5060,
SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
-- SIP/216.52.153.207-1418 answered SIP/217.168.168.49:5060
-- Attempting native bridge of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-1418
WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt):
Maximum retries exceeded on call
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] for seqno 1 (Response)
  == Spawn extension (default, 1303, 1) exited non-zero on
'SIP/217.168.168.49:5060'
WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt):
Maximum retries exceeded on call
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] for seqno 102 (Request)
-- Executing Dial(SIP/217.168.168.49:5060,
SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
-- SIP/216.52.153.207-11ed answered SIP/217.168.168.49:5060
-- Attempting native bridge of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-11ed
WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt):
Maximum retries exceeded on call
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] for seqno 1 (Response)
  == Spawn extension

Re: [Asterisk-Users] a beginner's SIP question .. (further to previous mailing)

2003-06-03 Thread Dave Alan Caruana




-- Executing Dial("SIP/sipphone-b6e6", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] -- 
SIP/216.52.153.207-ab35 answered SIP/sipphone-b6e6 -- 
Attempting native bridge of SIP/sipphone-b6e6 and 
SIP/216.52.153.207-ab35

is what shows up on the console window 
...

thanks again :)
Dave


  - Original Message - 
  From: 
  Dan 
  To: [EMAIL PROTECTED] 
  
  Sent: Friday, May 30, 2003 7:50 PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  Hi Dave,
  
  If you have registered theSIP phone with 
  Asterisk, then you must have a line like:
  
  exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
  
  in extensions.conf file
  
  Then call 555 from the SIP phone to access the 
  destination.
  
  BR,
  Dan
  
- Original Message - 
From: 
Dave Alan 
Caruana 
To: [EMAIL PROTECTED] 

Sent: Friday, May 30, 2003 6:21 
PM
Subject: Re: [Asterisk-Users] a 
beginner's SIP question ..

I have included a dump of the debug info 
...
what I am trying to do is route a call from 
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway 
[EMAIL PROTECTED]
If i dial direct from the sip phone to the 
gateway it works fine .. so 
I do not think there is any incompatibility 
there.
Calls don't go through though ...

please help!!!

cheers
Dave


*CLI -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-eca2 answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2WARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-1418 answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418WARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-11ed answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-11edWARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request)

  - Original Message - 
  From: 
  Dan 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, May 29, 2003 8:15 
  PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  Hi,
  
  
  Check to have a common set of 
  codecs.
  If X-Lite is used and at the other end is a 
  phone without GSM support, then it doesn't work.
  Try to disable GSM on the soft phone (if 
  X-Lite).
  
  BR,
  Dan
  
  
  
- Original Message - 
From: 
Dave Alan 
Caruana 
To: [EMAIL PROTECTED] 

Sent: Thursday, May 29, 2003 9:01 
PM
Subject: [Asterisk-Users] a 
beginner's SIP question ..

I am trying to get asterisk to dial this 
address :
sip:[EMAIL PROTECTED]

Using a softphone on my PC 
(217.168.168.49)
it dials immediately and I get a voice 
prompt ..

I have configured an extension, 1303 on 
asterisk,
modifying the demo configuration 
:

exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])

When from my softphone I dial
sip:[EMAIL PROTECTED]

on the console I get :
 -- Executing 
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.20

Re: [Asterisk-Users] a beginner's SIP question .. (further!)

2003-06-03 Thread Dave Alan Caruana




more about the same problem ...
i've been playing around and got to this error 
message which seems relevant ..

*CLI dial 1303 -- 
Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] -- 
SIP/216.52.153.207-1fb9 answered OSS/dsp Console call has been 
answered NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): 
Unknown RTP codec 19 receivedNOTICE[1232188736]: File rtp.c, Line 326 
(ast_rtp_read): Unknown RTP codec 19 receivedNOTICE[1232188736]: File rtp.c, 
Line 326 (ast_rtp_read): Unknown RTP codec 19 receivedNOTICE[1232188736]: 
File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 
receivedKilled
am I right in thinking i need a different codec to 
connect to the sip host I want to
connect to? where do codecs come from?

many cheers
Dave


  - Original Message - 
  From: 
  Dan 
  To: [EMAIL PROTECTED] 
  
  Sent: Friday, May 30, 2003 7:50 PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  Hi Dave,
  
  If you have registered theSIP phone with 
  Asterisk, then you must have a line like:
  
  exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
  
  in extensions.conf file
  
  Then call 555 from the SIP phone to access the 
  destination.
  
  BR,
  Dan
  
- Original Message - 
From: 
Dave Alan 
Caruana 
To: [EMAIL PROTECTED] 

Sent: Friday, May 30, 2003 6:21 
PM
Subject: Re: [Asterisk-Users] a 
beginner's SIP question ..

I have included a dump of the debug info 
...
what I am trying to do is route a call from 
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway 
[EMAIL PROTECTED]
If i dial direct from the sip phone to the 
gateway it works fine .. so 
I do not think there is any incompatibility 
there.
Calls don't go through though ...

please help!!!

cheers
Dave


*CLI -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-eca2 answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2WARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-1418 answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418WARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-11ed answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-11edWARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request)

  - Original Message - 
  From: 
  Dan 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, May 29, 2003 8:15 
  PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  Hi,
  
  
  Check to have a common set of 
  codecs.
  If X-Lite is used and at the other end is a 
  phone without GSM support, then it doesn't work.
  Try to disable GSM on the soft phone (if 
  X-Lite).
  
  BR,
  Dan
  
  
  
- Original Message - 
From: 
Dave Alan 
Caruana 
To: [EMAIL PROTECTED] 

Sent: Thursday, May 29, 2003 9:01 
PM
Subject: [Asterisk-Users] a 
beginner's SIP question ..

I am trying to get asterisk to dial this 
address :
sip:[EMAIL PROTECTED]

Using a softphone on my PC 
(217.168.168.49)
it dials immediately and I get a voice 
prom

Re: [Asterisk-Users] a beginner's SIP question ..

2003-06-03 Thread Dave Alan Caruana




sorry i'm sending so many emails, I always think of 
something
exactly after i've pressed Send .. please be 
patient with me :)

I also have OH323 installed, supposedly correctly, 
and the same
gateway I want to connect to on SIP also supports 
H323, however
i do not know what the dialcommand line for 
H323 is .. i'm trying

exten = 1304,1,Dial(OH323/216.52.153.206) 
;ring
but I actually want to dial extension 723 on the 
remote end,
so this is surely not right.. current messages i'm 
getting
from Asterisk are these :

*CLI dial 1304 -- 
Executing Dial("OSS/dsp", "OH323/216.52.153.206") in new 
stack*CLI 
0:03.623 
H323 Cleaner H323 Connection ip$localhost/9771 
terminated.ERROR[1232188736]: File chan_oh323.c, Line 610 (oh323_call): 
H323:0: Could not call 216.52.153.206. -- Couldn't call 
216.52.153.206 -- Hungup 'H323:0' == Everyone is 
busy at this time
help *very* welcome ;)

cheers
Dave

  - Original Message - 
  From: 
  Dan 
  To: [EMAIL PROTECTED] 
  
  Sent: Friday, May 30, 2003 7:50 PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  Hi Dave,
  
  If you have registered theSIP phone with 
  Asterisk, then you must have a line like:
  
  exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
  
  in extensions.conf file
  
  Then call 555 from the SIP phone to access the 
  destination.
  
  BR,
  Dan
  
- Original Message - 
From: 
Dave Alan 
Caruana 
To: [EMAIL PROTECTED] 

Sent: Friday, May 30, 2003 6:21 
PM
Subject: Re: [Asterisk-Users] a 
beginner's SIP question ..

I have included a dump of the debug info 
...
what I am trying to do is route a call from 
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway 
[EMAIL PROTECTED]
If i dial direct from the sip phone to the 
gateway it works fine .. so 
I do not think there is any incompatibility 
there.
Calls don't go through though ...

please help!!!

cheers
Dave


*CLI -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-eca2 answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2WARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-1418 answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418WARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-11ed answered 
SIP/217.168.168.49:5060 -- Attempting native bridge of 
SIP/217.168.168.49:5060 and SIP/216.52.153.207-11edWARNING[1125329600]: 
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) 
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File 
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request)

  - Original Message - 
  From: 
  Dan 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, May 29, 2003 8:15 
  PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  Hi,
  
  
  Check to have a common set of 
  codecs.
  If X-Lite is used and at the other end is a 
  phone without GSM support, then it doesn't work.
  Try to disable GSM on the soft phone (if 
  X-Lite).
  
  BR,
  Dan
  
  
  
- Original Message - 
From: 
Dave Alan 
Caruana 
To: [EMAIL PROTECTED] 

Sent: Thursday, May 29, 2003 9:01 
PM
Subject: [Asterisk-Users] a 
beginner's SIP question ..

I am trying to get asterisk to dial this 
address :
sip:[EMAIL PROTECTED]

Using a

Re: [Asterisk-Users] a beginner's SIP question ..

2003-05-31 Thread Dave Alan Caruana




I have included a dump of the debug info 
...
what I am trying to do is route a call from 
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway [EMAIL PROTECTED]
If i dial direct from the sip phone to the gateway 
it works fine .. so 
I do not think there is any incompatibility 
there.
Calls don't go through though ...

please help!!!

cheers
Dave


*CLI -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] -- 
SIP/216.52.153.207-eca2 answered SIP/217.168.168.49:5060 
-- Attempting native bridge of SIP/217.168.168.49:5060 and 
SIP/216.52.153.207-eca2WARNING[1125329600]: File chan_sip.c, Line 404 
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited 
non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c, 
Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] -- 
SIP/216.52.153.207-1418 answered SIP/217.168.168.49:5060 
-- Attempting native bridge of SIP/217.168.168.49:5060 and 
SIP/216.52.153.207-1418WARNING[1125329600]: File chan_sip.c, Line 404 
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited 
non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c, 
Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request) -- Executing 
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] -- 
SIP/216.52.153.207-11ed answered SIP/217.168.168.49:5060 
-- Attempting native bridge of SIP/217.168.168.49:5060 and 
SIP/216.52.153.207-11edWARNING[1125329600]: File chan_sip.c, Line 404 
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited 
non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c, 
Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request)

  - Original Message - 
  From: 
  Dan 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, May 29, 2003 8:15 
PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  Hi,
  
  
  Check to have a common set of 
codecs.
  If X-Lite is used and at the other end is a phone 
  without GSM support, then it doesn't work.
  Try to disable GSM on the soft phone (if 
  X-Lite).
  
  BR,
  Dan
  
  
  
- Original Message - 
From: 
Dave Alan 
Caruana 
To: [EMAIL PROTECTED] 

Sent: Thursday, May 29, 2003 9:01 
PM
Subject: [Asterisk-Users] a beginner's 
SIP question ..

I am trying to get asterisk to dial this 
address :
sip:[EMAIL PROTECTED]

Using a softphone on my PC 
(217.168.168.49)
it dials immediately and I get a voice prompt 
..

I have configured an extension, 1303 on 
asterisk,
modifying the demo configuration :

exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])

When from my softphone I dial
sip:[EMAIL PROTECTED]

on the console I get :
 -- Executing 
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new 
stack -- Called [EMAIL PROTECTED] 
-- SIP/216.52.153.207-7c3b answered SIP/sipphone-97b6 
-- Attempting native bridge of SIP/sipphone-97b6 and 
SIP/216.52.153.207-7c3b

but on my headset all I get is silence .. the 
call doesn't drop though.

What am I doing wrong ?

many thanks,
Dave

Re: [Asterisk-Users] a beginner's SIP question ..

2003-05-31 Thread Dan




Hi Dave,

If you have registered theSIP phone with 
Asterisk, then you must have a line like:

exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)

in extensions.conf file

Then call 555 from the SIP phone to access the 
destination.

BR,
Dan

  - Original Message - 
  From: 
  Dave Alan Caruana 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Friday, May 30, 2003 6:21 PM
  Subject: Re: [Asterisk-Users] a 
  beginner's SIP question ..
  
  I have included a dump of the debug info 
  ...
  what I am trying to do is route a call from 
  sipphone 217.168.168.49
  through asterisk 217.168.168.51 onto a gateway [EMAIL PROTECTED]
  If i dial direct from the sip phone to the 
  gateway it works fine .. so 
  I do not think there is any incompatibility 
  there.
  Calls don't go through though ...
  
  please help!!!
  
  cheers
  Dave
  
  
  *CLI -- Executing 
  Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
  stack -- Called [EMAIL PROTECTED] 
  -- SIP/216.52.153.207-eca2 answered 
  SIP/217.168.168.49:5060 -- Attempting native bridge of 
  SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2WARNING[1125329600]: 
  File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
  for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited 
  non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c, 
  Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
  for seqno 1 (Response) -- Executing 
  Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
  stack -- Called [EMAIL PROTECTED] 
  -- SIP/216.52.153.207-1418 answered 
  SIP/217.168.168.49:5060 -- Attempting native bridge of 
  SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418WARNING[1125329600]: 
  File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
  for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited 
  non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c, 
  Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
  for seqno 102 (Request) -- Executing 
  Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new 
  stack -- Called [EMAIL PROTECTED] 
  -- SIP/216.52.153.207-11ed answered 
  SIP/217.168.168.49:5060 -- Attempting native bridge of 
  SIP/217.168.168.49:5060 and SIP/216.52.153.207-11edWARNING[1125329600]: 
  File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
  for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited 
  non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c, 
  Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] 
  for seqno 102 (Request)
  
- Original Message - 
From: 
Dan 
To: [EMAIL PROTECTED] 

    Sent: Thursday, May 29, 2003 8:15 
    PM
Subject: Re: [Asterisk-Users] a 
beginner's SIP question ..

Hi,


Check to have a common set of 
codecs.
If X-Lite is used and at the other end is a 
phone without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if 
X-Lite).

BR,
Dan



  - Original Message - 
  From: 
  Dave Alan 
  Caruana 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, May 29, 2003 9:01 
  PM
  Subject: [Asterisk-Users] a 
  beginner's SIP question ..
  
  I am trying to get asterisk to dial this 
  address :
  sip:[EMAIL PROTECTED]
  
  Using a softphone on my PC 
  (217.168.168.49)
  it dials immediately and I get a voice prompt 
  ..
  
  I have configured an extension, 1303 on 
  asterisk,
  modifying the demo configuration 
  :
  
  exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])
  
  When from my softphone I dial
  sip:[EMAIL PROTECTED]
  
  on the console I get :
   -- Executing 
  Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new 
  stack -- Called [EMAIL PROTECTED] 
  -- SIP/216.52.153.207-7c3b answered 
  SIP/sipphone-97b6 -- Attempting native bridge of 
  SIP/sipphone-97b6 and SIP/216.52.153.207-7c3b
  
  but on my headset all I get is silence .. the 
  call doesn't drop though.
  
  What am I doing wrong ?
  
  many thanks,
  Dave

Re: [Asterisk-Users] a beginner's SIP question ..

2003-05-30 Thread Dan




Hi,


Check to have a common set of codecs.
If X-Lite is used and at the other end is a phone 
without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if 
X-Lite).

BR,
Dan



  - Original Message - 
  From: 
  Dave Alan Caruana 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, May 29, 2003 9:01 
PM
  Subject: [Asterisk-Users] a beginner's 
  SIP question ..
  
  I am trying to get asterisk to dial this address 
  :
  sip:[EMAIL PROTECTED]
  
  Using a softphone on my PC 
  (217.168.168.49)
  it dials immediately and I get a voice prompt 
  ..
  
  I have configured an extension, 1303 on 
  asterisk,
  modifying the demo configuration :
  
  exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])
  
  When from my softphone I dial
  sip:[EMAIL PROTECTED]
  
  on the console I get :
   -- Executing 
  Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new 
  stack -- Called [EMAIL PROTECTED] 
  -- SIP/216.52.153.207-7c3b answered SIP/sipphone-97b6 -- 
  Attempting native bridge of SIP/sipphone-97b6 and 
  SIP/216.52.153.207-7c3b
  
  but on my headset all I get is silence .. the 
  call doesn't drop though.
  
  What am I doing wrong ?
  
  many thanks,
  Dave

Re: [Asterisk-Users] a beginner's SIP question ..

Re: [Asterisk-Users] a beginner's SIP question .. (further to previous mailing)

Re: [Asterisk-Users] a beginner's SIP question .. (further!)

Re: [Asterisk-Users] a beginner's SIP question ..

Re: [Asterisk-Users] a beginner's SIP question ..

Re: [Asterisk-Users] a beginner's SIP question ..

Re: [Asterisk-Users] a beginner's SIP question ..

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