[asterisk-users] SIP client open URL upon answer (was : Re: SIP client able to handle Access-URL: header)

[Bertrand LUPART - Linkeo.com]

Hello,


> I'm playing with the optional URL parameter of the Dial() command, which 
> "will also be sent to the called party upon successful connection, if the 
> channel technology supports the sending of URLs in this way."[1]
> 
> Basically, the following asterisk dialplan directive :
> 
> - - 8< - - 8< - - 8< - -
> same => n,Dial(SIP/1234,30,tr,http://www.example.com/)
> - - >8 - - >8 - - >8 - -
> 
> Will generate the following SIP header in the INVITE request :
> 
> - - 8< - - 8< - - 8< - -
> Access-URL: ;mode=active
> - - >8 - - >8 - - >8 - -
> 
> Does anyone know about a SIP client able to handle this header ? I'd expect 
> it to automatically open this webpage.

Self-answering after some research on this topic.

Seems like "Access-URL" isn't an official SIP header :
http://www.iana.org/assignments/sip-parameters/sip-parameters.xhtml#sip-parameters-2

This Dial() optional URL parameter looks like the same beast than SendURL() 
added nearly 10 years ago in asterisk SIP stack :
https://github.com/asterisk/asterisk/commit/04e45cfda35bd3cb4e508aad9286b5deb99ccf56

Is anyone aware of a SIP client which would open an URL sent by asterisk?


Thank you,

-- 
Bertrand LUPART

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[asterisk-users] SIP client able to handle Access-URL: header

[Bertrand LUPART - Linkeo.com]

Hello,


I'm playing with the optional URL parameter of the Dial() command, which "will 
also be sent to the called party upon successful connection, if the channel 
technology supports the sending of URLs in this way."[1]

Basically, the following asterisk dialplan directive :

- - 8< - - 8< - - 8< - -
same => n,Dial(SIP/1234,30,tr,http://www.example.com/)
- - >8 - - >8 - - >8 - -

Will generate the following SIP header in the INVITE request :

- - 8< - - 8< - - 8< - -
Access-URL: ;mode=active
- - >8 - - >8 - - >8 - -


Does anyone know about a SIP client able to handle this header ? I'd expect it 
to automatically open this webpage.


Thank you,


[1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

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[asterisk-users] Sip-Client / type=peer / Why can this client place calls?

[Thorsten Göllner]

Hi,

I am using Asterisk 11.5.1. As far as I understood, the following 
configuration allows a sip client only to receive calls (type=peer) but 
not to place calls 
(http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place 
calls though with this config?


sip.conf
...
[thorsten]
type=peer
host=dynamic
context=my_context
nat=force_rport,comedia
secret=...
dtmfmode=rfc2833
disallow=all
allow=g722
allow=g729
allow=g729
...

extensions.conf
...
[my_context]
exten = _X.,1,Dial(DAHDI/g1/${EXTEN},60)
...

Of course: when removing a valid context the client can not place the 
call. But I thought this behaviour can be controlled via type=peer?!


Thanks in advance
-Thorsten-



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Re: [asterisk-users] Sip-Client / type=peer / Why can this client place calls?

[Steve Totaro]

On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com wrote:

 Hi,

 I am using Asterisk 11.5.1. As far as I understood, the following
 configuration allows a sip client only to receive calls (type=peer) but not
 to place calls 
 (http://www.voip-info.org/**wiki/view/Asterisk+sip+typehttp://www.voip-info.org/wiki/view/Asterisk+sip+type).
 Why can I place calls though with this config?

 sip.conf
 ...
 [thorsten]
 type=peer
 host=dynamic
 context=my_context
 nat=force_rport,comedia
 secret=...
 dtmfmode=rfc2833
 disallow=all
 allow=g722
 allow=g729
 allow=g729
 ...

 extensions.conf
 ...
 [my_context]
 exten = _X.,1,Dial(DAHDI/g1/${EXTEN},**60)
 ...

 Of course: when removing a valid context the client can not place the
 call. But I thought this behaviour can be controlled via type=peer?!

 Thanks in advance
 -Thorsten-


See if this is helpful.

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
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Re: [asterisk-users] Sip-Client / type=peer / Why can this client place calls?

[Thorsten Göllner]

Thanks a lot. Seems to be a good hidden page, isn't it? ;-)

Am 03.09.2013 14:30, schrieb Steve Totaro:




On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com 
mailto:t...@ovm-group.com wrote:


Hi,

I am using Asterisk 11.5.1. As far as I understood, the following
configuration allows a sip client only to receive calls
(type=peer) but not to place calls
(http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I
place calls though with this config?

sip.conf
...
[thorsten]
type=peer
host=dynamic
context=my_context
nat=force_rport,comedia
secret=...
dtmfmode=rfc2833
disallow=all
allow=g722
allow=g729
allow=g729
...

extensions.conf
...
[my_context]
exten = _X.,1,Dial(DAHDI/g1/${EXTEN},60)
...

Of course: when removing a valid context the client can not place
the call. But I thought this behaviour can be controlled via
type=peer?!

Thanks in advance
-Thorsten-


See if this is helpful.

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer


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[asterisk-users] SIP client that supports T.38 Fax

[Ahmed Munir]

Hi,

I'm looking for SIP client that supports T.38 Fax other than zoiper.

Please advise at earliest.

-- 
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Ahmed Munir Chohan
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[asterisk-users] SIP client on a mobile?

[Per Jessen]

Steve Totaro wrote:

 Just use a SIP client on your phone.  Many providers have multiple
 failover paths for inbound calls.
 
 This thread morphed from a nice home phone system into something
 completely different.

Yup.

  For my situation, DISA is pointless except for road warriors who
  call all over the world, from anywhere, they can call into the corp
  system, get dialtone and skip the whole process of expense reports
  for work
  related calls.  It makes things less complex, not more.

 Using DISA also means getting a corp caller id, not a mobile.
 
 Yes, spoofing provides that.
 

  Maybe if you explain your situation and how your plan works, but
  for me, personally, DISA would be a an added cost and complication.
 
  The only purpose I can think of for myself could be accomplished by
  spoofing caller id.

 How is that done from a mobile?  Sofar that has been my main reason
 for using DISA - cost is not a real issue.
 
 SIP client.  Spoof card, yes it is DISA, but you don't have to do
 anything but use the card.

Steve, even if I could get SIP clients for our phones, doesn't this mean
using a data connection rather than just voice?  That would make it a
lot pricier than the current setup with DISA (which is largely free).


/Per Jessen, Zürich

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Re: [asterisk-users] SIP client on a mobile?

[Steve Totaro]

On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote:

 Steve Totaro wrote:

  Just use a SIP client on your phone.  Many providers have multiple
  failover paths for inbound calls.
 
  This thread morphed from a nice home phone system into something
  completely different.

 Yup.

   For my situation, DISA is pointless except for road warriors who
   call all over the world, from anywhere, they can call into the corp
   system, get dialtone and skip the whole process of expense reports
   for work
   related calls.  It makes things less complex, not more.
 
  Using DISA also means getting a corp caller id, not a mobile.
 
  Yes, spoofing provides that.
 
 
   Maybe if you explain your situation and how your plan works, but
   for me, personally, DISA would be a an added cost and complication.
  
   The only purpose I can think of for myself could be accomplished by
   spoofing caller id.
 
  How is that done from a mobile?  Sofar that has been my main reason
  for using DISA - cost is not a real issue.
 
  SIP client.  Spoof card, yes it is DISA, but you don't have to do
  anything but use the card.

 Steve, even if I could get SIP clients for our phones, doesn't this mean
 using a data connection rather than just voice?  That would make it a
 lot pricier than the current setup with DISA (which is largely free).


 /Per Jessen, Zürich


A Wifi connection?  I guess that wifi is not like it is here.  I can get on
highspeed wifi anywhere I go in the DC Metro area for free.  Just driving
around, there is always an open access point.  When driving around, I pick
up thousands of APs in a couple miles and don't have any protection at all.

I would suspect that most road warriors have high speed data needs?  Not
sure what business you are in, but having fast internet (relatively
speaking) is a must to do work.  I am not saying to use the data supplied
from phone, if that is what you are thinking.

If your phones don't have SIP, then use callback.  You call your company, go
through whatever you seutp in the dialplan, and the phone system calls you
back as well as calling the other party.

You edited out much of the context of the conversation to support your
side.  I don't play games like that...

SIP client on the phone was an option.  Was the original question about
using DISA to save money?  Yes it was.  Now you are stating that it is
largely free.

Callback is a great solution when outbound cell phone calls quite a bit more
than your cutrate VoIP provider.  As I said, many countries do not charge
for inbound calls.

I am still clueless what your point is/was but if it is almost free then,
stick with it.  Still clueless why you posted if it almost free.

Thanks,
Steve Totaro
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Re: [asterisk-users] SIP client on a mobile?

[Steve Totaro]

On Thu, Aug 25, 2011 at 3:06 AM, Steve Totaro 
stot...@totarotechnologies.com wrote:



 On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote:

 Steve Totaro wrote:

  Just use a SIP client on your phone.  Many providers have multiple
  failover paths for inbound calls.
 
  This thread morphed from a nice home phone system into something
  completely different.

 Yup.

   For my situation, DISA is pointless except for road warriors who
   call all over the world, from anywhere, they can call into the corp
   system, get dialtone and skip the whole process of expense reports
   for work
   related calls.  It makes things less complex, not more.
 
  Using DISA also means getting a corp caller id, not a mobile.
 
  Yes, spoofing provides that.
 
 
   Maybe if you explain your situation and how your plan works, but
   for me, personally, DISA would be a an added cost and complication.
  
   The only purpose I can think of for myself could be accomplished by
   spoofing caller id.
 
  How is that done from a mobile?  Sofar that has been my main reason
  for using DISA - cost is not a real issue.
 
  SIP client.  Spoof card, yes it is DISA, but you don't have to do
  anything but use the card.

 Steve, even if I could get SIP clients for our phones, doesn't this mean
 using a data connection rather than just voice?  That would make it a
 lot pricier than the current setup with DISA (which is largely free).


 /Per Jessen, Zürich


 A Wifi connection?  I guess that wifi is not like it is here.  I can get on
 highspeed wifi anywhere I go in the DC Metro area for free.  Just driving
 around, there is always an open access point.  When driving around, I pick
 up thousands of APs in a couple miles and don't have any protection at all.

 I would suspect that most road warriors have high speed data needs?  Not
 sure what business you are in, but having fast internet (relatively
 speaking) is a must to do work.  I am not saying to use the data supplied
 from phone, if that is what you are thinking.

 If your phones don't have SIP, then use callback.  You call your company,
 go through whatever you seutp in the dialplan, and the phone system calls
 you back as well as calling the other party.

 You edited out much of the context of the conversation to support your
 side.  I don't play games like that...

 SIP client on the phone was an option.  Was the original question about
 using DISA to save money?  Yes it was.  Now you are stating that it is
 largely free.

 Callback is a great solution when outbound cell phone calls quite a bit
 more than your cutrate VoIP provider.  As I said, many countries do not
 charge for inbound calls.

 I am still clueless what your point is/was but if it is almost free then,
 stick with it.  Still clueless why you posted if it almost free.

 Thanks,
 Steve Totaro


I am not sure why people try to prove me wrong, but they do.  On rare
occasions, I am wrong, I am also big enough to admit it.

To answer your question, and get on the same terms, VoIP (or data as you
prefer) would probably be cheaper.  Isn't that the whole reason behind
VoIP?  You say voice, does that mean your provider's voice service?

Depending on the cost of inbound and out abound calls on a cell are the key
here.

Is it next to nothing to call a foreign country from your cell?  Is it much
more expensive than rates at the office.  Generally, I think outbound calls
from an office are much lower than cell phone charges.

I was paying a $40k plus weekly for long distance calls from Iraq to mostly
Fiji, Uganda, Peru.  That was with VoicePulse, all 703 DIDs around the
world.  Voicepulse gave me great rates because $40k a week is not chump
change.  I wonder what the cost of cell phone calls would amount to?

My international rates for outbound cell phone calls are beyond a rip-off.

Thanks,
Steve Totaro
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Re: [asterisk-users] SIP client on a mobile?

[Per Jessen]

Steve Totaro wrote:

 Steve, even if I could get SIP clients for our phones, doesn't this
 mean using a data connection rather than just voice?  That would make
 it a lot pricier than the current setup with DISA (which is largely
 free). 


 /Per Jessen, Zürich


 A Wifi connection?  I guess that wifi is not like it is here.  I can
 get on highspeed wifi anywhere I go in the DC Metro area for free. 

In the cities, WiFi is typically only available in restaurants and cafes
(Starbucks, McDonalds etc).  In the country, no wifi.  Well, the odd
open access point, but using it is illegal, so that's a no-go.

 I would suspect that most road warriors have high speed data needs? 
 Not sure what business you are in, but having fast internet
 (relatively speaking) is a must to do work.  I am not saying to use
 the data supplied from phone, if that is what you are thinking.

For my company, the mobile is primarily for voice - people don't spend
that much time on the road, but when they do, they still want to appear
as if they're in the office. 

 If your phones don't have SIP, then use callback.  You call your
 company, go through whatever you seutp in the dialplan, and the phone
 system calls you back as well as calling the other party.
 
 You edited out much of the context of the conversation to support your
 side.  I don't play games like that...

Sorry, that wasn't my intention, I just snip out the bits that aren't
relevant to a reply. 

 SIP client on the phone was an option.  Was the original question
 about using DISA to save money?  Yes it was.  Now you are stating that
 it is largely free.

I think the OPs question was about saving money, to which I suggested
using DISA - it my setup it's largely free. 

 Callback is a great solution when outbound cell phone calls quite a
 bit more than your cutrate VoIP provider.  As I said, many countries
 do not charge for inbound calls.

Right.

 I am still clueless what your point is/was but if it is almost free
 then, stick with it.  Still clueless why you posted if it almost free.

I did not post the original question, I just responded to it.


/Per Jessen, Zürich

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Re: [asterisk-users] SIP client floods port 5060 and gets blocked

[Julian Yap]

On Thu, Oct 28, 2010 at 1:42 AM, Jonas Kellens jonas.kell...@telenet.bewrote:

 On 10/28/2010 12:52 PM, Gordon Henderson wrote:
  On Thu, 28 Oct 2010, Jonas Kellens wrote
  On 10/28/2010 10:44 AM, Kevin Keane wrote:
 
  I assume that you checked and the remote IP is a legitimate IP phone?
 If
  not, it could be an attempt to break into your system.
 
  If it is a legitimate IP phone, make sure that the SIP configuration is
  correct -- if the SIP authentication fails, you can see this happening.
 
 
  1. This is a legitimate phone, yes.
  2. Registration goes as follow : REGISTER  SIP/2.0 401 Unauthorized
  Re-Register with Digest  200 OK
 
  Is it s Snom phone?
 
  I've seen Snoms do this...
 
  Gordon
 

 I have this with Snom 320, Snom 370, Grandstream GXW4008 and YeaLink T28...


Yes, I have seen this with Snom 370s...  It's maddening.  I'm going to start
testing out the version 8.x firmware.

- Julian
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Re: [asterisk-users] SIP client floods port 5060 and gets blocked

[Jonas Kellens]

Hello,

any more input on this subject ?!


Kind regards,
Jonas.


 Original Message 
Subject:Re: [asterisk-users] SIP client floods port 5060 and gets 
blocked
Date:   Thu, 28 Oct 2010 13:42:12 +0200
From:   Jonas Kellens jonas.kell...@telenet.be
To: 	Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com




On 10/28/2010 12:52 PM, Gordon Henderson wrote:

 On Thu, 28 Oct 2010, Jonas Kellens wrote

 On 10/28/2010 10:44 AM, Kevin Keane wrote:


 I assume that you checked and the remote IP is a legitimate IP phone? If
 not, it could be an attempt to break into your system.

 If it is a legitimate IP phone, make sure that the SIP configuration is
 correct -- if the SIP authentication fails, you can see this happening.



 1. This is a legitimate phone, yes.
 2. Registration goes as follow : REGISTER   SIP/2.0 401 Unauthorized
 Re-Register with Digest   200 OK


 Is it s Snom phone?

 I've seen Snoms do this...

 Gordon



I have this with Snom 320, Snom 370, Grandstream GXW4008 and YeaLink T28...


Jonas.

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[asterisk-users] SIP client floods port 5060 and gets blocked

[Jonas Kellens]

Hello,

Is there any reason why an IP-phone would pounder on port 5060 ? My 
firewall blocks the public IP because it thinks the remote IP is port 
scanning on port 5060.


I think the phone is just registering but for some reason it does this 
repeatedly in a very short time.



Oct 28 09:01:48 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48073 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:01:49 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48074 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:01:50 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48075 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:01:52 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48076 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:01:56 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48077 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:02:00 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48078 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:02:04 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48079 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:02:08 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48083 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:02:12 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48084 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:02:16 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48085 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676
Oct 28 09:02:20 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip 
DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48087 DF PROTO=UDP 
SPT=2367 DPT=5060 LEN=676



Any input on this ?!


Kind regards,
Jonas.
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Re: [asterisk-users] SIP client floods port 5060 and gets blocked

[Kevin Keane]

I assume that you checked and the remote IP is a legitimate IP phone? If not, 
it could be an attempt to break into your system.

If it is a legitimate IP phone, make sure that the SIP configuration is correct 
- if the SIP authentication fails, you can see this happening.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, October 28, 2010 12:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client floods port 5060 and gets blocked

Hello,

Is there any reason why an IP-phone would pounder on port 5060 ? My firewall 
blocks the public IP because it thinks the remote IP is port scanning on port 
5060.

I think the phone is just registering but for some reason it does this 
repeatedly in a very short time.


Oct 28 09:01:48 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48073 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:01:49 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48074 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:01:50 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48075 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:01:52 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48076 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:01:56 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48077 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:02:00 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48078 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:02:04 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48079 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:02:08 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48083 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:02:12 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48084 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:02:16 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48085 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676
Oct 28 09:02:20 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= 
MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip 
LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48087 DF PROTO=UDP SPT=2367 DPT=5060 
LEN=676


Any input on this ?!


Kind regards,
Jonas.
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Re: [asterisk-users] SIP client floods port 5060 and gets blocked

[Jonas Kellens]

On 10/28/2010 10:44 AM, Kevin Keane wrote:


I assume that you checked and the remote IP is a legitimate IP phone? 
If not, it could be an attempt to break into your system.


If it is a legitimate IP phone, make sure that the SIP configuration 
is correct -- if the SIP authentication fails, you can see this happening.




1. This is a legitimate phone, yes.
2. Registration goes as follow : REGISTER  SIP/2.0 401 Unauthorized  
Re-Register with Digest  200 OK



Regards,
Jonas.
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Re: [asterisk-users] SIP client floods port 5060 and gets blocked

[Gordon Henderson]

On Thu, 28 Oct 2010, Jonas Kellens wrote:

 On 10/28/2010 10:44 AM, Kevin Keane wrote:
 
 I assume that you checked and the remote IP is a legitimate IP phone? If 
 not, it could be an attempt to break into your system.
 
 If it is a legitimate IP phone, make sure that the SIP configuration is 
 correct -- if the SIP authentication fails, you can see this happening.
 

 1. This is a legitimate phone, yes.
 2. Registration goes as follow : REGISTER  SIP/2.0 401 Unauthorized  
 Re-Register with Digest  200 OK

Is it s Snom phone?

I've seen Snoms do this...

Gordon

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Re: [asterisk-users] SIP client floods port 5060 and gets blocked

[Jonas Kellens]

On 10/28/2010 12:52 PM, Gordon Henderson wrote:
 On Thu, 28 Oct 2010, Jonas Kellens wrote
 On 10/28/2010 10:44 AM, Kevin Keane wrote:
  
 I assume that you checked and the remote IP is a legitimate IP phone? If
 not, it could be an attempt to break into your system.

 If it is a legitimate IP phone, make sure that the SIP configuration is
 correct -- if the SIP authentication fails, you can see this happening.


 1. This is a legitimate phone, yes.
 2. Registration goes as follow : REGISTER  SIP/2.0 401 Unauthorized
 Re-Register with Digest  200 OK
  
 Is it s Snom phone?

 I've seen Snoms do this...

 Gordon


I have this with Snom 320, Snom 370, Grandstream GXW4008 and YeaLink T28...


Jonas.

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[asterisk-users] SIP client MAC address.

[DHAVAL INDRODIYA]

hello,

is there any facility to get SIP client (ex. softphone,ipphone) MAC address
on asterisk.

based on that we authenticated client in anyway.

i tried with sip debug but i didn't got any MAC address related field in all
packets.



regards
Dhaval
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Re: [asterisk-users] SIP client MAC address.

[Klaverstyn, David C]

From Linux you could use

 

arp | grep 192.168.0.1

 

substituting the IP address of the SIP device.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.

 

hello,

is there any facility to get SIP client (ex. softphone,ipphone) MAC
address on asterisk.

based on that we authenticated client in anyway.

i tried with sip debug but i didn't got any MAC address related field in
all packets.



regards
Dhaval

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Re: [asterisk-users] SIP client MAC address.

[DHAVAL INDRODIYA]

hello david,
what in case of sip client is behind NAT, and i want SIP client IP
address. not from system from which client
registered.  if it is a SIP phone then what? if you have any idea then tell
me.

regards
dhaval

On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C 
david.klavers...@intergraph.com wrote:

  From Linux you could use



 arp | grep 192.168.0.1



 substituting the IP address of the SIP device.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
 *Sent:* Wednesday, 28 October 2009 4:29 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] SIP client MAC address.



 hello,

 is there any facility to get SIP client (ex. softphone,ipphone) MAC address
 on asterisk.

 based on that we authenticated client in anyway.

 i tried with sip debug but i didn't got any MAC address related field in
 all packets.



 regards
 Dhaval

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Re: [asterisk-users] SIP client MAC address.

[Klaverstyn, David C]

If there is more than one SIP devices operating from the same NAT device
then I'm not sure what you could do as it would always show the same IP
for all SIP devices behind the same NAT.  If there is only one device
behind that NAT making a connection to your server then that is easy, if
not I think your screwed.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP client MAC address.

 

hello david,
what in case of sip client is behind NAT, and i want SIP client IP
address. not from system from which client
registered.  if it is a SIP phone then what? if you have any idea then
tell me.

regards 
dhaval

On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C
david.klavers...@intergraph.com wrote:

From Linux you could use

 

arp | grep 192.168.0.1

 

substituting the IP address of the SIP device.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.

 

hello,

is there any facility to get SIP client (ex. softphone,ipphone) MAC
address on asterisk.

based on that we authenticated client in anyway.

i tried with sip debug but i didn't got any MAC address related field in
all packets.



regards
Dhaval


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Re: [asterisk-users] SIP client MAC address.

[DHAVAL INDRODIYA]

hi,

though , the SIP client is behinf the NAT cannot we get MAC address of that
client , from SIP headers.
or do you suggest any alternate method .

regards
dhaval

On Wed, Oct 28, 2009 at 12:20 PM, Klaverstyn, David C 
david.klavers...@intergraph.com wrote:

  If there is more than one SIP devices operating from the same NAT device
 then I’m not sure what you could do as it would always show the same IP for
 all SIP devices behind the same NAT.  If there is only one device behind
 that NAT making a connection to your server then that is easy, if not I
 think your screwed.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
 *Sent:* Wednesday, 28 October 2009 4:47 PM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] SIP client MAC address.



 hello david,
 what in case of sip client is behind NAT, and i want SIP client IP
 address. not from system from which client
 registered.  if it is a SIP phone then what? if you have any idea then tell
 me.

 regards
 dhaval

 On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C 
 david.klavers...@intergraph.com wrote:

 From Linux you could use



 arp | grep 192.168.0.1



 substituting the IP address of the SIP device.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
 *Sent:* Wednesday, 28 October 2009 4:29 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] SIP client MAC address.



 hello,

 is there any facility to get SIP client (ex. softphone,ipphone) MAC address
 on asterisk.

 based on that we authenticated client in anyway.

 i tried with sip debug but i didn't got any MAC address related field in
 all packets.



 regards
 Dhaval


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Re: [asterisk-users] SIP client MAC address.

[Alex Balashov]

This is a very strange discussion.

MAC addresses can only be discovered for peers that are on the same 
broadcast segment - which is the realm within which ARP lookups 
participate.

Any peers not on the same logical Layer 2 network are reached through 
a Layer 3 hop.  MAC addresses behind that routing hop cannot be found 
out because the nodes are in a different MAC domain.

NAT has absolutely nothing to do with this, and thus is irrelevant one 
way or another.

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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[asterisk-users] SIP client Resp code

[DHAVAL INDRODIYA]

hello,

I have SIP phone registered with my server

now if they send me any number for dialing then i want to give a response
code

actually this number is conference number and i need to chek via DB query
that this conference is valid or not

if conference is not valid then i want to send a response code to SIP device
that conf not valid

i tried with Sendtext but not getting success


here are my dialplan

exten = _X.,1,NOOP(${CALLERID(num)}${CALLERID(DNID)}--)
exten = _X.,n,SendText(Hello You are Now connected)
exten = _X.,n,SipAddheader(RESP_CODE : Hello You are Now connected)
exten = _X.,n,Meetme(151515,sdMAwC)
exten = _X.,n,Hangup()

can anybody help this out
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Re: [asterisk-users] Sip client registers then unregisters

[Dovid B]

Have you tried using another phone and compare the results ?

- Original Message - 
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, May 16, 2007 2:34 PM
Subject: [asterisk-users] Sip client registers then unregisters


I have a remote user with Eyebeam on a laptop. Internet connectivity seems 
good, there is no packet loss to that location from the PBX.
Everytime the user starts eyebeam, the application tries to register. 
Asterisk accepts the registration but the reply never gets to the client 
application, so it thinks it has not been accepted and times out. Then 
Asterisk unregisters the extension.


   -- Registered SIP '881' at 212.248.xxx.xxx port 26605 expires 300
   -- Unregistered SIP '881'

Anyone got any ideas how to debug and fix this?

--
Chris Mason



--
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[asterisk-users] Sip client registers then unregisters

[Chris Mason (Lists)]

I have a remote user with Eyebeam on a laptop. Internet connectivity 
seems good, there is no packet loss to that location from the PBX.
Everytime the user starts eyebeam, the application tries to register. 
Asterisk accepts the registration but the reply never gets to the client 
application, so it thinks it has not been accepted and times out. Then 
Asterisk unregisters the extension.


   -- Registered SIP '881' at 212.248.xxx.xxx port 26605 expires 300
   -- Unregistered SIP '881'

Anyone got any ideas how to debug and fix this?

--
Chris Mason



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Re: [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?

[Tim Panton]

On 5 Oct 2006, at 21:05, Joe wrote:


Actually the xclients could run the SIP clients but they have firewall
restrictions.


Use IAX  on the xclients (I'm semi-serious)

Tim Panton

www.mexuar.com



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Re: [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?

[Tzafrir Cohen]

On Thu, Oct 05, 2006 at 03:05:40PM -0500, Joe wrote:
 Actually the xclients could run the SIP clients but they have firewall
 restrictions.
 
 I want to SSH to the machines which aren't behind the firewall and
 pull the SIP client interfaces back via X Windows.

ssh is a very poor choice for such a firewall.

Read http://tldp.org/HOWTO/ppp-ssh/introduction.html#DRAWBACKS

Now re-read the recent thread we had here about the extra overhead of
RTP packets. With an ssh tunnel you'll typically get a data packet of
512 bytes for every RTP packet. Nice overhead.

Consider using openvpn or something similar instead.


Also: You did not mention where the the X clients reside. Maybe your
task is actually to work with a decent networked sound server just as
display is networked with X.

Some interesting, but quite linux-specific resources regarding that may
be found at http://wiki.ltsp.org/ . 

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?

[Joe]

Actually the xclients could run the SIP clients but they have firewall
restrictions.

I want to SSH to the machines which aren't behind the firewall and
pull the SIP client interfaces back via X Windows.

On 10/4/06, Tim Panton [EMAIL PROTECTED] wrote:


On 4 Oct 2006, at 18:35, Joe wrote:

 Hello,

 I'm looking for a SIP client that work with Asterisk that will run on
 Linux or Solaris and will work with X Windows. I know X won't all the
 media to work but I'm really only interested in SIP signaling.

What are you running your X displays on ?
You may find that they are capable of running a local
softphone.

I had our java IAX client working on solaris - the media stream had
quite a bit of latency, but I figure with some tuning it could be
got working well.

Tim Panton

www.mexuar.com



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[asterisk-users] SIP client that runs on Linux or Solaris through X Windows?

[Joe]

Hello,

I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.

Thanks!
Joe
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Re: [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?

[Tim Panton]

On 4 Oct 2006, at 18:35, Joe wrote:


Hello,

I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.


What are you running your X displays on ?
You may find that they are capable of running a local
softphone.

I had our java IAX client working on solaris - the media stream had
quite a bit of latency, but I figure with some tuning it could be
got working well.

Tim Panton

www.mexuar.com



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Re: [asterisk-users] SIP client with video???

[Joao Pereira]

The problems with X-Lite 3 are:

- just accepts one SIP registration
- doesnt send video to other X-Lite or eyeBeam versions
- sometimes loses the SIP informations when you reboot the PC
.
Regards
Joao Pereira

Blake Krone wrote:

What's wrong with X-Lite 3.0? I haven't had any issues with it and 
find it to be one of the best SIP video software choices, and it's free.


On 7/27/06, *Joao Pereira * [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hello to all
can someone recommend me a nice SIP client with video for windows??

I tried X-Lite 3.0 but it's a lousy piece of software.

Does someone knows about a better software?
Regards
Joao Pereira

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Re: [asterisk-users] SIP client with video???

[Blake Krone]

I've never experienced any of those problems. I can send video to other eyeBeam versions, 3.0 is the only version that supports video on the lite side. I've never lost any SIP information and only one registration isn't a big deal to me. If you need more than one buy the full eyebeam version.
On 9/6/06, Joao Pereira [EMAIL PROTECTED] wrote:
The problems with X-Lite 3 are:- just accepts one SIP registration- doesnt send video to other X-Lite or eyeBeam versions- sometimes loses the SIP informations when you reboot the PC.Regards
Joao PereiraBlake Krone wrote: What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free. On 7/27/06, *Joao Pereira * 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello to all can someone recommend me a nice SIP client with video for windows??
 I tried X-Lite 3.0 but it's a lousy piece of software. Does someone knows about a better software? Regards Joao Pereira ___
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Re: [asterisk-users] SIP client with video???

[richard Coco]

Hi,

i have xlite too and it works without any problems.

ps: what about ekiga? (www.ekiga.org)

rich

--- Joao Pereira [EMAIL PROTECTED] wrote:

 Hello to all
 can someone recommend me a nice SIP client with
 video for windows??
 
 I tried X-Lite 3.0 but it's a lousy piece of
 software.
 
 Does someone knows about a better software?
 Regards
 Joao Pereira
 
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[asterisk-users] SIP client with video???

[Joao Pereira]

Hello to all
can someone recommend me a nice SIP client with video for windows??

I tried X-Lite 3.0 but it's a lousy piece of software.

Does someone knows about a better software?
Regards
Joao Pereira

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Re: [asterisk-users] SIP client with video???

[Blake Krone]

What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free.On 7/27/06, Joao Pereira
 [EMAIL PROTECTED] wrote:Hello to all
can someone recommend me a nice SIP client with video for windows??I tried X-Lite 3.0 but it's a lousy piece of software.Does someone knows about a better software?RegardsJoao Pereira
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[Asterisk-Users] SIP client looses register and then i need to restart my pc to get registered on Asterisk 1.2.5

[Marco Mouta]

Hi all,

I've a some users on my network, reporting this:

Sjphone is registered , and some times just looses registry in
Asterisk, I don't know if it is expiration ( instead of loosing
registry).

Then to get registered again they need to restart their own PC.

Why could this beeing happening?

Best regards,
Marco Mouta
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[Asterisk-Users] Sip Client

[bodra]

-- Original Message --
From: bodra [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date:  Sun, 28 Aug 2005 02:35:01 -0700

Hi all

 i am developing a client for the asterisk that controls ur phone from an Xp c# 
application

what functions in Asterisk that will allow you to put someone on hold but not 
park calls and bring them back, without using flash hook cuz it will be a 
button in that application and i think i couldnt send a flash hook signal to 
the server..



Regards
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[Asterisk-Users] Sip client

[gale81]

Hello!

If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale



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Re: [Asterisk-Users] Sip client

[Dave Walker]

[EMAIL PROTECTED] wrote:


Hello!

If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale



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Have you tried JAIN?
http://www.google.co.uk/search?q=sip+java+api
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[Asterisk-Users] SIP Client at outside and connect to an Asterisk Server sit behind NAT with SER

[Stephen Liew]

Hi ,
I have setup SER and Asterisk on the samebox (behind a NAT) and try to 
connect to SER from outside using Xlite.
But I got 405 error.

Here is my setup diagram.
   NAT
SIP Client - Internet (PPPoE ---Router ---SER+Asterisk)
 Same Network
Can anyone please highlight to me where went wrong?
I am using dynamic DNS for my SER + Asterisk and using a dynamic DNS 
domain name for my SIP Client.

--  
Thanks and Regards,
Stephen Liew
begin:vcard
fn:Stephen Liew
n:Liew;Stephen
adr:Tmn Melodies;;14A, Jln Geronggang;Johor Bahru;Johor;80250;Malaysia
email;internet:[EMAIL PROTECTED]
tel;work:+(60) 7 334 9781
tel;fax:+(60) 7 334 5502
tel;cell:+(60)-12-7107350
url:http://www.revoltel.com
version:2.1
end:vcard

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[Asterisk-Users] SIP client cannot connect to Asterisk

[K Wong]

Hi:

We have got SIP clients connecting to our Asterisk fine with a DSL
connection behind router (NAT), but when we bring the Sipura 2000 ATA
to a Rogers Cable connection behind a Netgear router (NAT), the SIP
clients aren't able to reach the Asterisk at all.

We enabled the SIP debug in Asterisk, and it doesn't see any request
coming from these SIP clients, and we also tried the to use a XTEN
Lite to connect to Simpletelecom within this network and it fails to
register as well.

It seems to be a network configuration problem, but there isn't much
log in the router that we can dignose as the Netgear router WR814 only
logs TCP web requests.

Does anyone know what the problem could be?  Does Rogers Cable blocks SIP ports?
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RE: [Asterisk-Users] SIP client cannot connect to Asterisk

[Bill Hamlin]

Try setting the SIP signalling port in your client to something other than
5060 (eg ) and run tethereal on your Asterisk box to see if you're
getting packets on .


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of K Wong
Sent: Monday, December 27, 2004 9:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP client cannot connect to Asterisk


Hi:

We have got SIP clients connecting to our Asterisk fine with a DSL
connection behind router (NAT), but when we bring the Sipura 2000 ATA
to a Rogers Cable connection behind a Netgear router (NAT), the SIP
clients aren't able to reach the Asterisk at all.

We enabled the SIP debug in Asterisk, and it doesn't see any request
coming from these SIP clients, and we also tried the to use a XTEN
Lite to connect to Simpletelecom within this network and it fails to
register as well.

It seems to be a network configuration problem, but there isn't much
log in the router that we can dignose as the Netgear router WR814 only
logs TCP web requests.

Does anyone know what the problem could be?  Does Rogers Cable blocks SIP
ports?
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Re: [Asterisk-Users] SIP client cannot connect to Asterisk

[hanson]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
put this in sip.conf
;externip = x.x.x.x; Address that we're going to put in outbound
SIP messages ( official ip address)
;localnet=10.0.0.0/255.255.255.0; if we're behind a NAT
add
nat=yes to every sip account which is behind of NAT
BR Hanson
K Wong wrote:
| Hi:
|
| We have got SIP clients connecting to our Asterisk fine with a DSL
| connection behind router (NAT), but when we bring the Sipura 2000
| ATA to a Rogers Cable connection behind a Netgear router (NAT), the
| SIP clients aren't able to reach the Asterisk at all.
|
| We enabled the SIP debug in Asterisk, and it doesn't see any
| request coming from these SIP clients, and we also tried the to use
| a XTEN Lite to connect to Simpletelecom within this network and it
| fails to register as well.
|
| It seems to be a network configuration problem, but there isn't
| much log in the router that we can dignose as the Netgear router
| WR814 only logs TCP web requests.
|
| Does anyone know what the problem could be?  Does Rogers Cable
| blocks SIP ports? ___
| Asterisk-Users mailing list Asterisk-Users@lists.digium.com
| http://lists.digium.com/mailman/listinfo/asterisk-users To
| UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
|
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFB0Cm675fQW47llQgRAltKAJwNIEdzKBc14V8VgH/dU2HziXE/fgCfRUjr
3YvQZP+V8rYMm+uKlaiLOvg=
=RbSJ
-END PGP SIGNATURE-
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Re: [Asterisk-Users] SIP Client for Symbian

[Roy Sigurd Karlsbakk]

Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
well
running GSM or some fancy codec over GPRS or UMTS may well make sense :)
roy
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RE: [Asterisk-Users] SIP Client for Symbian

[Noah Miller]

Hi Dean -
Noah, what client were you using on your treo for this 600ms voip call?
Oh, I wasn't using a SIP client (is there one for palm?).  Sorry if 
that was misleading - this is just web browsing and email.  Once the 
connection gets going, it is able to do the 2.2 KB/s that standard GPRS 
should be able to do, but every time it has to initiate a new 
connection it's like hitting a brick wall - it takes forever.  Very 
high latency.  Maybe I'll get some network tools and measure it 
sometime.

Thanks,
Noah
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Re: [Asterisk-Users] SIP Client for Symbian

[Dinesh Nair]

On 09/12/2004 05:54 Steven Critchfield said the following:
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
the nokia 9500 (communicator) as sold in asia uses symbian and has built-in 
802.11b. i can see a software SIP phone here being useful for corporate 
types, can't you ? :)

see http://www.nokia.com.my/nokia/0,,57140,00.html
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|   for b in clients employers associates relatives neighbours pets; do   |
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[Asterisk-Users] SIP Client for Symbian

[Andre Remor]

Hi,

I'm looking for a SIP client for Symbian 
OS...
Someone known one? (free or not)

Tks

Andre Remor
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
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Re: [Asterisk-Users] SIP Client for Symbian

[Steven Critchfield]

On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
 Hi,
  
 I'm looking for a SIP client for Symbian OS...
 Someone known one? (free or not)

Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] SIP Client for Symbian

[Paul Rodan]

Sprint's PTT uses SIP. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, December 08, 2004 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Client for Symbian

On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
 Hi,
  
 I'm looking for a SIP client for Symbian OS...
 Someone known one? (free or not)

Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] SIP Client for Symbian

[dean collins]

Steven, I think it makes total sense. I'm currently in the process of
trying to source a better solution than my treo 600 in order to stay in
contact (looking at either ppc or xp with cf/gsm adaptor)

Anything that will enable me to answer my home and office and
international office lines from a single device using sip/iax over gprs
would be a boon in both time and energy and money saving.

It's no longer your grandfather's telephone network.


Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, December 08, 2004 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Client for Symbian

On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
 Hi,
  
 I'm looking for a SIP client for Symbian OS...
 Someone known one? (free or not)

Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] SIP Client for Symbian

[Peter Svensson]

On Wed, 8 Dec 2004, Steven Critchfield wrote:

 On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
  I'm looking for a SIP client for Symbian OS...
  Someone known one? (free or not)
 
 Unless Symbian has branched off of cell phones, I doubt it. SIP on a
 cell phone right now doesn't make sense.

There are Symbian phones with wlan, such as the Nokia 9500. There is a SIP 
stack in some Symbian phones (if they are new enough). This is needed 
anyway since the PTT stuff seems to be sip based.

Also, Symbian used to be used in the Psion palmtops and then migrated to 
mobile phones when Ericsson, Nokia and Motorola bought a large chunk of 
the operation from Psion and formed Symbian.

Peter


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Re: [Asterisk-Users] SIP Client for Symbian

[Steve Kennedy]

On Wed, Dec 08, 2004 at 03:54:10PM -0600, Steven Critchfield wrote:

 On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
  Hi,
  I'm looking for a SIP client for Symbian OS...
  Someone known one? (free or not)
 Unless Symbian has branched off of cell phones, I doubt it. SIP on a
 cell phone right now doesn't make sense.

Err, 3G should use SIP. Nextel's push-to-talk service uses SIP.

Skype (though not SIP) works on PocketPC's/smartphones, I'm sure there's
a SIP client out there ...


Steve

-- 
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RE: [Asterisk-Users] SIP Client for Symbian

[Steven Critchfield]

On Wed, 2004-12-08 at 17:16 -0500, dean collins wrote:
 Steven, I think it makes total sense. I'm currently in the process of
 trying to source a better solution than my treo 600 in order to stay in
 contact (looking at either ppc or xp with cf/gsm adaptor)
 
 Anything that will enable me to answer my home and office and
 international office lines from a single device using sip/iax over gprs
 would be a boon in both time and energy and money saving.
 
 It's no longer your grandfather's telephone network.

Maybe you should experience what is available in the states. I don't see
enough bandwidth nor a reasonable cost just yet in gprs service to
justify it over plain old celular time. 

For instance, my GPRS is billed at 1 cent per K or $10.24 per meg. Can
you see how quickly a VoIP call would rack up the price of the call?
Most cell plans offer free nights and weekends for limited to no extra
cost. Data keeps costing no matter the time. Also most cell plans are
under 10 cents per minute without needing anything other than a standard
phone on the other side.

As for the few plans I have seen around for $80 a month to get unlimited
network on the phones, I haven't seen that they are capable of the speed
needed to get the calls through it. And again, at 10 cents a minute,
thats a lot of time calling on the phone.
-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] SIP Client for Symbian

[Noah Miller]

I'm looking for a SIP client for Symbian OS...
Someone known one? (free or not)
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.

Steven, I think it makes total sense. I'm currently in the process of
trying to source a better solution than my treo 600 in order to stay in
contact (looking at either ppc or xp with cf/gsm adaptor)

Anything that will enable me to answer my home and office and
international office lines from a single device using sip/iax over gprs
would be a boon in both time and energy and money saving.
It's no longer your grandfather's telephone network.
I don't think Steven really meant that it doesn't make sense.  I think 
he meant that it isn't practical.  I've read that latency on GPRS 
connections (even on the faster ATT EDGE and Sprint VISION networks) 
is somewhere around 600 ms.  I've never personally measured it, but I 
have used a Treo 600 on the T-Mobile GPRS network, and I can tell you 
that that latency is WAY too high to make any real phone calls.  200 ms 
is about the maximum latency you can really deal with on a VoIP call, 
but even that is really too high.  At 600 ms, your voice wouldn't get 
to the other party forever, and vice-versa.  It might be useful as a 
Nextel-like walkie talkie device, but a real phone conversation would 
be painful.

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Re: [Asterisk-Users] SIP Client for Symbian

[tim panton]

Take a look a JSR 180 for MIDP Cellphones.
google: j2me sip jsr 180
finds lots of links,
including a Nokia reference implementation.
But from what I can see it only addresses the SIP protocol itself,
not the media streams or how to play them.
Tim.
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RE: [Asterisk-Users] SIP Client for Symbian

[dean collins]

Noah, what client were you using on your treo for this 600ms voip call?


Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Wednesday, December 08, 2004 5:50 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP Client for Symbian

 I'm looking for a SIP client for Symbian OS...
 Someone known one? (free or not)

 Unless Symbian has branched off of cell phones, I doubt it. SIP on a
 cell phone right now doesn't make sense.

 Steven, I think it makes total sense. I'm currently in the process of
 trying to source a better solution than my treo 600 in order to stay
in
 contact (looking at either ppc or xp with cf/gsm adaptor)

 Anything that will enable me to answer my home and office and
 international office lines from a single device using sip/iax over
gprs
 would be a boon in both time and energy and money saving.

 It's no longer your grandfather's telephone network.

I don't think Steven really meant that it doesn't make sense.  I think 
he meant that it isn't practical.  I've read that latency on GPRS 
connections (even on the faster ATT EDGE and Sprint VISION networks) 
is somewhere around 600 ms.  I've never personally measured it, but I 
have used a Treo 600 on the T-Mobile GPRS network, and I can tell you 
that that latency is WAY too high to make any real phone calls.  200 ms 
is about the maximum latency you can really deal with on a VoIP call, 
but even that is really too high.  At 600 ms, your voice wouldn't get 
to the other party forever, and vice-versa.  It might be useful as a 
Nextel-like walkie talkie device, but a real phone conversation would 
be painful.

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RE: [Asterisk-Users] SIP Client for Symbian

[dean collins]

I'm in NY (was in Australia) I have flat rate GPRS and utilize it for
streaming audio on a regular basis (shoutcasts etc)

$49 a month flat rate gprs through ATT

No if I was still in Sydney I'd continue to use my www.iburst.com.au it
just came on the market as I left, wireless dsl speeds, covers almost
all of the Sydney region with Canberra this year and 5 other cities to
follow next year.

No that with the new sony U71 cant be beat.



Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, December 08, 2004 5:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP Client for Symbian

On Wed, 2004-12-08 at 17:16 -0500, dean collins wrote:
 Steven, I think it makes total sense. I'm currently in the process of
 trying to source a better solution than my treo 600 in order to stay
in
 contact (looking at either ppc or xp with cf/gsm adaptor)
 
 Anything that will enable me to answer my home and office and
 international office lines from a single device using sip/iax over
gprs
 would be a boon in both time and energy and money saving.
 
 It's no longer your grandfather's telephone network.

Maybe you should experience what is available in the states. I don't see
enough bandwidth nor a reasonable cost just yet in gprs service to
justify it over plain old celular time. 

For instance, my GPRS is billed at 1 cent per K or $10.24 per meg. Can
you see how quickly a VoIP call would rack up the price of the call?
Most cell plans offer free nights and weekends for limited to no extra
cost. Data keeps costing no matter the time. Also most cell plans are
under 10 cents per minute without needing anything other than a standard
phone on the other side.

As for the few plans I have seen around for $80 a month to get unlimited
network on the phones, I haven't seen that they are capable of the speed
needed to get the calls through it. And again, at 10 cents a minute,
thats a lot of time calling on the phone.
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] SIP Client for Symbian

[Jon Radon]

I think it's about 600ms under ideal conditions.  I normally see
around 1000ms or higher on Tmo GPRS.  EDGE will not aid in latency. 
We'll have to wait for UMTS with possible HSDPA to see around 200ms,
or so I hear.


On Wed, 8 Dec 2004 17:49:58 -0500, Noah Miller [EMAIL PROTECTED] wrote:
 I don't think Steven really meant that it doesn't make sense.  I think
 he meant that it isn't practical.  I've read that latency on GPRS
 connections (even on the faster ATT EDGE and Sprint VISION networks)
 is somewhere around 600 ms.  I've never personally measured it, but I
 have used a Treo 600 on the T-Mobile GPRS network, and I can tell you
 that that latency is WAY too high to make any real phone calls.  200 ms
 is about the maximum latency you can really deal with on a VoIP call,
 but even that is really too high.  At 600 ms, your voice wouldn't get
 to the other party forever, and vice-versa.  It might be useful as a
 Nextel-like walkie talkie device, but a real phone conversation would
 be painful.

-- 
Is it something someone said, was it something someone said?
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[Asterisk-Users] SIP client registration ignored by Asterisk

[Chris Johnson]

I have an xlite client registering with asterisk. The client is on a
separate subnet routed via a site to site VPN tunnel to the subnet
hosting the * server, with no active firewall between the subnets.

With one IP address (192.168.3.56) set on the client the * box ignores
the SIP REGISTER request. 

Ethereal logs on both sides of the link show the REGISTER packets
arriving at the * box, but no response. With the problem IP address the
client can PING, SSH and Telnet to the * manager port successfully and
can connect to the web service on the * box.

Changing the IP address (192.168.3.99) of the client immediately lets
the Registration complete successfully.

The problem has persisted through several reboots and through a
stop/start of the Ethernet interface.

Verbose asterisk output (level 50) and sip debug shows no activity
whilst the packets are ignored, and a search of /var/logs/asterisk files
gives nothing to indicate a problem

Debug output from the xlite client (below) illustrates the problem and
the immediate connection upon changing client IP address.

I have reviewed previous postings but cannot find any pointer to the
root cause of this.

Does anyone have any previous similar experience or thoughts as to the
possible cause?

Xlite output:
SEND TIME: 145834312
SEND  192.168.2.10:5060
REGISTER sip:asterisk.chersoft.local SIP/2.0
Via: SIP/2.0/UDP
192.168.3.56:5060;rport;branch=z9hG4bK07FE6665CB4D4E649D564DDF78ACBAA9
From: ChrisJ sip:[EMAIL PROTECTED];tag=742343849
To: ChrisJ sip:[EMAIL PROTECTED]
Contact: ChrisJ sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 60736 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


SEND TIME: 145837562
SEND  192.168.2.10:5060
REGISTER sip:asterisk.chersoft.local SIP/2.0
Via: SIP/2.0/UDP
192.168.3.56:5060;rport;branch=z9hG4bK07FE6665CB4D4E649D564DDF78ACBAA9
From: ChrisJ sip:[EMAIL PROTECTED];tag=742343849
To: ChrisJ sip:[EMAIL PROTECTED]
Contact: ChrisJ sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 60736 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


SEND TIME: 145843812
SEND  192.168.2.10:5060
REGISTER sip:asterisk.chersoft.local SIP/2.0
Via: SIP/2.0/UDP
192.168.3.56:5060;rport;branch=z9hG4bK07FE6665CB4D4E649D564DDF78ACBAA9
From: ChrisJ sip:[EMAIL PROTECTED];tag=742343849
To: ChrisJ sip:[EMAIL PROTECTED]
Contact: ChrisJ sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 60736 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0

Attempting SIP protocol listen on: 192.168.3.99:5060

Established SIP protocol listen on: 192.168.3.99:5060

SIP: 192.168.3.99:5060
RTP: 192.168.3.99:8000
NAT: 111.222.333.444

Discovered Blocked Firewall


SEND TIME: 145913312
SEND  192.168.2.10:5060
REGISTER sip:asterisk.chersoft.local SIP/2.0
Via: SIP/2.0/UDP
192.168.3.99:5060;rport;branch=z9hG4bK37A5C1D916484A8392C38E75BA948677
From: ChrisJ sip:[EMAIL PROTECTED];tag=742343849
To: ChrisJ sip:[EMAIL PROTECTED]
Contact: ChrisJ sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 60737 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 145913390
RECEIVE  192.168.2.10:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.3.99:5060;branch=z9hG4bK37A5C1D916484A8392C38E75BA948677
From: ChrisJ sip:[EMAIL PROTECTED];tag=742343849
To: ChrisJ sip:[EMAIL PROTECTED];tag=as4c6fbb1b
Call-ID: [EMAIL PROTECTED]
CSeq: 60737 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


RECEIVE TIME: 145913406
RECEIVE  192.168.2.10:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.3.99:5060;branch=z9hG4bK37A5C1D916484A8392C38E75BA948677
From: ChrisJ sip:[EMAIL PROTECTED];tag=742343849
To: ChrisJ sip:[EMAIL PROTECTED];tag=as4c6fbb1b
Call-ID: [EMAIL PROTECTED]
CSeq: 60737 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 1800
Contact: sip:[EMAIL PROTECTED]:5060;expires=1800
Date: Tue, 30 Nov 2004 09:35:15 GMT
Content-Length: 0


RECEIVE TIME: 145919703
RECEIVE  192.168.2.10:5060
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK13ebbee2
From: asterisk sip:[EMAIL PROTECTED];tag=as2041a68b
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/2

SEND TIME: 145919718
SEND  192.168.2.10:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK13ebbee2
From: asterisk sip:[EMAIL PROTECTED];tag=as2041a68b
To: sip:[EMAIL PROTECTED]:5060;tag=2331908490
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
Server: X-Lite release 1103m
Content-Length: 0

Chris Johnson
CherSoft Ltd

Re: [Asterisk-Users] SIP client to IAXTel 800/888/877/866 dialing thru Asterisk

[Jason Williams]

On Mon, 12 Jul 2004 11:42:57 -0700, Dameon D. Welch-Abernathy
[EMAIL PROTECTED] wrote:
 Through my Asterisk server, I am trying to use IAXTel to dial 800-type
 numbers (yes, I know I can do the same thing with FWD and others via
 SIP, but I wanted to play with IAX a little). It appears I'm running
 into some sort of a codec mismatch or something because it's not working
 right. The SIP client is a SPA-3000.
 
Phoneboy

IAXcomm use gsm only that may help

Jason
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[Asterisk-Users] SIP client to IAXTel 800/888/877/866 dialing thru Asterisk

[Dameon D. Welch-Abernathy]

Through my Asterisk server, I am trying to use IAXTel to dial 800-type
numbers (yes, I know I can do the same thing with FWD and others via
SIP, but I wanted to play with IAX a little). It appears I'm running
into some sort of a codec mismatch or something because it's not working
right. The SIP client is a SPA-3000. 

In iax.conf, I have something like the following:

[General]
disallow=all
allow=ilbc
allow=adpcm
allow=g726
allow=gsm
allow=ulaw
allow=alaw

register = user:[EMAIL PROTECTED]

[iaxtel]
type=friend
host=iaxtel.com
secret=password
auth=rsa
context=from-iaxtel
inkeys=iaxtel

The sip.conf has similar settings in the General section related to
codecs. The codec on the SPA-3000 is forced to G.711 (the only thing it
talks to directly is my Asterisk server). 

When I make a call to a number on IAXTel (e.g. 1-700-VOXILLA), it works
fine. If I try and call an 800 number, I don't get any audio. However,
it appears the IAX side of the equation isn't negotiating a codec right.
Furthermore, the SIP side is choosing the wrong codec (though I've tried
putting allow=ulaw first and it still didn't help)

grover*CLI iax2 show channels
Peer UsernameID (Lo/Rem)  Seq (Tx/Rx)  Lag  Jitter 
JitBuf  Format
69.73.19.178 (None)  3/0  1/0  0ms  ms 
ms  UNKN
69.73.19.178 phoneboy5/00102  00019/00017  00099ms  ms 
0010ms  UNKN
2 active IAX channel(s)
grover*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)   Format
10.0.0.250   53  2b890d18-49  00101/00103   ILBC
1 active SIP channel(s)

Is there a problem with iaxtel? Any ideas?

Asterisk CVS-HEAD-07/06/04-01:33:49 built by [EMAIL PROTECTED] on a i686
running Linux

-- PhoneBoy

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[Asterisk-Users] sip client

[Altus Snyman]

Good day.
I'm looking for a sip client 4 fedora???
Frdora?
Thanks
Altus

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[Asterisk-Users] SIP client auth

[George Lin]

Dear list,

does any one know how to do a SIP client auth via central database instead
of specifying in the sip.conf ?

if we could do with central database, should we use RADIUS or other better
way to do it.

Thanks,

George


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RE: [Asterisk-Users] SIP-Client for Handheld PC

[Hans-Henrik Andresen]

Hi,

Yes Telesym, xten and one more I can't remember the name of it, they are all 
for PPC-only. :(

/HHA


From: Ray Burkholder [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP-Client for Handheld PC
Date: Mon, 12 Jan 2004 14:33:39 -0500
What are the ones you found for PocketPC?  I guess you've looked at the
Telesym site?  They have a SIP flavor coming out shortly for some PDA's.
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 644 6999 x2002
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Hans-Henrik Andresen
 Sent: January 12, 2004 05:01
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] SIP-Client for Handheld PC


 Anyone know a sip-client that will work on a Handheld PC
 running WINCE for
 HPC.

 I can find some for PocketPC, but the wont work on my HPC

 ??

 /HHA
_
Let the new MSN Premium Internet Software make the most of your high-speed 
experience. http://join.msn.com/?pgmarket=en-uspage=byoa/premST=1

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[Asterisk-Users] SIP-Client for Handheld PC

[Hans-Henrik Andresen]

Anyone know a sip-client that will work on a Handheld PC running WINCE for 
HPC.

I can find some for PocketPC, but the wont work on my HPC

??

/HHA

_
Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! 
  http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1

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RE: [Asterisk-Users] SIP-Client for Handheld PC

[Ray Burkholder]

What are the ones you found for PocketPC?  I guess you've looked at the
Telesym site?  They have a SIP flavor coming out shortly for some PDA's.

Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 644 6999 x2002


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Hans-Henrik Andresen
 Sent: January 12, 2004 05:01
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] SIP-Client for Handheld PC
 
 
 Anyone know a sip-client that will work on a Handheld PC 
 running WINCE for 
 HPC.
 
 I can find some for PocketPC, but the wont work on my HPC
 
 ??
 
 /HHA
 
 _
 Scope out the new MSN Plus Internet Software - optimizes 
 dial-up to the max! 
http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1
 
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 http://www.oneunified.net and is believed to be clean.
 


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RE: [Asterisk-Users] SIP-Client for Handheld PC

[Michael Graves]

Xten (www.xten.com) have a version of X-lite for PPC.

Michael

On Mon, 12 Jan 2004 14:33:39 -0500, Ray Burkholder wrote:

What are the ones you found for PocketPC?  I guess you've looked at the
Telesym site?  They have a SIP flavor coming out shortly for some PDA's.

Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 644 6999 x2002


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Hans-Henrik Andresen
 Sent: January 12, 2004 05:01
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] SIP-Client for Handheld PC
 
 
 Anyone know a sip-client that will work on a Handheld PC 
 running WINCE for 
 HPC.
 
 I can find some for PocketPC, but the wont work on my HPC
 
 ??
 
 /HHA
 
 _
 Scope out the new MSN Plus Internet Software - optimizes 
 dial-up to the max! 
http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1
 
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--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

If you hear a redneck exclaim, Hey, y'all, watch this!  
Stay out of his way. These are likely the last words he will ever say.
 
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RE: [Asterisk-Users] SIP-Client for Handheld PC

[Michael Graves]

On Mon, 12 Jan 2004 18:44:41 -0600, Michael Graves wrote:

Xten (www.xten.com) have a version of X-lite for PPC.

Michael

On Mon, 12 Jan 2004 14:33:39 -0500, Ray Burkholder wrote:

What are the ones you found for PocketPC?  I guess you've looked at the
Telesym site?  They have a SIP flavor coming out shortly for some PDA's.

Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 644 6999 x2002


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Hans-Henrik Andresen
 Sent: January 12, 2004 05:01
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] SIP-Client for Handheld PC
 
 
 Anyone know a sip-client that will work on a Handheld PC 
 running WINCE for 
 HPC.
 
 I can find some for PocketPC, but the wont work on my HPC
 

Y'know reflecting upon this, and giving consideration to my once pricey
Toshiba e740 PPC with wifi, I can't think of any good reason to use a
soft phone on a PPC. Most straight PDAs (as opposed to smart phones)
only provide earphone jacks and a tiny built-in microphone originally
intended for voice notes. Battery life with wifi enabled is not great.
Has anyone here tried using a pda as a SIP phone? With any dgtree of
satisfaction? Practicality?

Michael


--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

I am easily satisfied with the very best - Winston Churchill  
 
The questions arisen, is this a prison? Some say it is, but I say it isn't.
 - Ian Hunter
 
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[Asterisk-Users] SIP client not registering to *

[Chandra]

xlite saying login timed out. contact network admin.

how to get rid of this. * is not behind NAT.

also, the grandstream SIP phone also seems to fail to register. IAX phones
are all ok.

DIAX works fine


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Re: [Asterisk-Users] SIP client

[Rattana BIV]

Thanks very much !!

I thinks it could be very useful for me

Regards
Rattana
- Original Message - 
From: Peer Oliver schmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003 7:14 PM
Subject: Re: [Asterisk-Users] SIP client


 Christopher Stephens schrieb:
 
   Is there SIP client which work with Asterisk and can be embedded in a
   HTML page ?
  It may not be *exactly* what you're looking for, but try:
  http://fwd.pulver.com/callme.php?userid=411
 [..]
 
 Unfortunately this seem to work with Internet Explorer, only.
 
 rgds
 pos
 
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[Asterisk-Users] SIP client

[Rattana BIV]

hi everybody,

Is there SIP client which work with Asterisk and 
can be embedded in a HTML page ?


Thanks


Rattana

Re: [Asterisk-Users] SIP client

[Gary]

On Wed, 29 Oct 2003 09:58:28 +0100, Rattana BIV wrote:



hi everybody,

Is there SIP client which work with Asterisk and can be embedded in a HTML page ?


Thanks


Rattana


hehe, why use a SIP client, why not a client which does IAX and bury
that in a web page ?

(yes, I haven't answered the original question :-)

Gary
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Re: [Asterisk-Users] SIP client

[Christopher Stephens]

It may not be *exactly* what you're looking for, but try:

http://fwd.pulver.com/callme.php?userid=411

In examining the source, it seems you can put any SIP address, not just
FWD ones, though there doesn't seem to be any overt SIP registration
going on.


- Original message -
From: Rattana BIV [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Wed, 29 Oct 2003 09:58:28 +0100
Subject: [Asterisk-Users] SIP client



hi everybody,

Is there SIP client which work with Asterisk and can be embedded in a
HTML page ?


Thanks


Rattana
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Re: [Asterisk-Users] SIP client

[Peer Oliver schmidt]

Christopher Stephens schrieb:

 Is there SIP client which work with Asterisk and can be embedded in a
 HTML page ?
It may not be *exactly* what you're looking for, but try:
http://fwd.pulver.com/callme.php?userid=411
[..]

Unfortunately this seem to work with Internet Explorer, only.

rgds
pos
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Re: [Asterisk-Users] SIP client-NAT-Asterisk-NAT-SIP client. only works with canreinvite=no.

[WipeOut .]

Can anybody explain me what does canreinvite=yes really does?

Not sure how technical an answer you want becasue it look slike you know whats going on but as I unterstand it canreinvite=no tells the UA that reinvite is not supported and so causes all the RTP traffic to be routed via the * server.. I played with many nat settings and port forwarding settings and it ended up that canreinvite=no was the solution to my problems as well.. the downside is that it requires more bandwidth at the central site but the plus side is that it works through NAT..


Any ideas on the client A to C (same LAN, same NAT box, unique outside 
IP, same * server)?

Only thing that springs to mind is to install another * box internally and then use IAX to connect the internal * box to the external one.. then the internal phone will call each other without crossing the NAT..

Later..
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Re: [Asterisk-Users] SIP client-NAT-Asterisk-NAT-SIP client. only works with canreinvite=no.

[Alastair Maw]

WipeOut . wrote:

Any ideas on the client A to C (same LAN, same NAT box, unique
outside IP, same * server)?
Only thing that springs to mind is to install another * box
internally and then use IAX to connect the internal * box to the
external one.. then the internal phone will call each other without
crossing the NAT..
It shouldn't be *too* hard to change Asterisk such that it allows 
reinvites for particular netmasks. If you can ensure that your NAT 
clients are on different subnets, for example, this might be possible.

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Re: [Asterisk-Users] SIP client-NAT-Asterisk-NAT-SIP client. only works with canreinvite=no.

[austino]

 I have been trying to get SIP UA work with NAT but i have no been
successful has any one got  NATed ATA working(i.e an ATA witha private IP
working with NAT).
Asterisk registers the 192.168.0.3 Ip but no call go through at all,
infact there is no log of any call made on asterisk console.

can anyone please send me the sip.conf and ATA 186 configs of a NATed ATA
to working with *.
This what i have in my sip.conf

[]
type=friend
username=
transfer=yes
nat=yes
canreinvite=no
context=myata
host=dynamic
permit=0.0.0.0/0.0.0.0
accountcode=mi100

  ATA configs
IP=192.168.0.3
staticRoute=192.168.0.2
mask=255.255.255.0
dhcp=0
GkorProxy= (*'s public IP)
gateway= (*'s Public IP)
outbound Proxy=(*'s public IP)
NATIP= (host machine's Public IP)



On Thu, 11 Sep 2003, Jose Ildefonso Camargo Tolosa wrote:

 Hi!

 I have this configuration:

 SIP client A - NAT box A (real external IP) - Asterisk server (real
 IP) - (real external IP) NAT box B - SIP client B

 The echo test form any of the clients to the asterisk server is working
 just fine, even without canreinvite=no.

 When I try to call from SIP client A to B, wihtout the canreinvite=no in
 the sip.conf, the call doesn't even ring.

 Then I add the canreinvite=no to BOTH clients on the sip.conf, it starts
 to work.  The problem is that all voice data goes through my asterisk
 server, so the delay is longer.

 Also, this config doesn't work:

 SIP client A - NAT box A (real external IP, only one) - Asterisk
 server (real IP)
 SIP client C - NAT box A (real external IP, only one) - Asterisk
 server (real IP).

 When I try to call from A to C or C to A, the phone doesn't even ring,
 again, the echo test work just fine.

 SIP client A and SIP client C are in the same LAN, and both goes through
 NAT box A to the same asterisk server.

 In the case of clients A and C, the native bridge would be great,
 because it would save bandwith to both, my client, and me, and the voice
 delay would be almost nothing.

 My problem is: According to the data I got from the sip debug and the
 X-lite debug outputs, I don't see any reazon why the native bridge can't
 work, both clients gets different ports on the outside IP of the nat
 box, and that port is correctly recognized, and the reinvite packet is
 correctly sent.

 Can anybody explain me what does canreinvite=yes really does?

 Any ideas on the client A to C (same LAN, same NAT box, unique outside
 IP, same * server)?

 Thanks in advance,

 Sincerely,

 Ildefonso Camargo
 [EMAIL PROTECTED]


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