[Asterisk-Users] Echo on internal SIP

2005-03-31 Thread Philip Siegrist 
Hi All,

On my * server I am getting echo on internal SIP calls. I.E. Sip 2
Sip. Calls going over the T1 via the T100p are fine.

I have used ulaw and gsm, gsm has less echo but it is still noticable.
All phones are snom 190s.  Any ideas on what i can do to cancel this.

Thanks,
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[Asterisk-Users] Menu Selections Only Work Internally

2005-02-11 Thread Philip Siegrist 
All,

Funny problem. During my greating, the menu selections only work if
one calls from an internal sip line.  The greating plays for all
including calls over the t1. But pressing 9 for directory or any other
mapped button will only work if I call from inside. If I arrive to the
menu from an outside line SIP or POTS pressing the button does
nothing. Any ideas?

extensions.conf

--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)

[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
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Re: [Asterisk-Users] Menu Selections Only Work Internally

2005-02-11 Thread Philip Siegrist 
yes. it get's to the Menu prompt which is defined under [MainMenu].
The input buttons simply do not work.


On Fri, 11 Feb 2005 09:06:26 -0600, Jay Milk [EMAIL PROTECTED] wrote:
 Does your incoming context include the MainMenu?
 
  -Original Message-
  From: Philip Siegrist [mailto:[EMAIL PROTECTED]
  Sent: Friday, February 11, 2005 8:17 AM
  To: Asterisk-Users@lists.digium.com
  Subject: [Asterisk-Users] Menu Selections Only Work Internally
 
 
  All,
 
  Funny problem. During my greating, the menu selections only
  work if one calls from an internal sip line.  The greating
  plays for all including calls over the t1. But pressing 9 for
  directory or any other mapped button will only work if I call
  from inside. If I arrive to the menu from an outside line SIP
  or POTS pressing the button does nothing. Any ideas?
 
  extensions.conf
 
  --
  [MainMenu]
  exten=s,1,Answer
  exten=s,2,Wait(1)
  exten=s,3,Background(main-menu)
  exten=_3XX,1,Goto(sip,${EXTEN},1)
  exten=0,1,Goto(sip,301,1)
 
  [sip]
  ;Main Number
  exten = 300,1,Goto(MainMenu,s,1)
  --
 
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[Asterisk-Users] General Inbound Calls

2005-02-10 Thread Philip Siegrist 
Beating my head against the wall with this one.

I simply want to allow all sip calls from any providor to be accpeted
by asterisk. Can someone give me a sanity check on this?

Thanks.
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Re: [Asterisk-Users] General Inbound Calls

2005-02-10 Thread Philip Siegrist 
Thanks, but it still fails to authenticate the user of sip:[EMAIL PROTECTED]


On Thu, 10 Feb 2005 12:16:51 -0600, Eric Wieling [EMAIL PROTECTED] wrote:
 Philip Siegrist wrote:
 
  Beating my head against the wall with this one.
 
  I simply want to allow all sip calls from any providor to be accpeted
  by asterisk. Can someone give me a sanity check on this?
 
 I thought that was what the insecure=very option in [general] was for
 in sip.conf.  Calls would fall into the context= specified in [general].

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[Asterisk-Users] Sip providor reference in extentions.conf

2005-01-06 Thread Philip Siegrist 
Hi,

I am stuck. :(  I am attempting to use a 3rd party voip company for LD
calling by doing the following:

sip.conf

[newtel]
type=peer
host=sip1.newtel.net
nat=yes
disallow=all
allow=g729

extentions.conf:

[newtel]
exten = _81.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _81.,2,Congestion

In this case Newtel does not need authentication, It identifies by my
sending IP. The problem is  SIP/${EXTEN:[EMAIL PROTECTED] does not reference
the host line in sip.conf entry for newtel. Looking at the firewall it
tries to connect to uslec-66-255-165-3.cust.uslec.net.sip.

If i replace the SIP/${EXTEN:[EMAIL PROTECTED] with
SIP/${EXTEN:[EMAIL PROTECTED] it does correctly attempt to contact
this host. however, I need to set options in sip.conf such as g729
codec, so this does not resolve my problem.

If someone can help me I will buy you a pony.

Thanks
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Re: [Asterisk-Users] Sip providor reference in extentions.conf

2005-01-06 Thread Philip Siegrist 
On Thu, 6 Jan 2005 18:30:40 -0500, Philip Siegrist [EMAIL PROTECTED] wrote:
 On Thu, 06 Jan 2005 16:13:38 -0700, Ryan Courtnage [EMAIL PROTECTED] wrote:
  On Thu, 2005-06-01 at 17:56 -0500, Philip Siegrist wrote:
   Hi,
  
   I am stuck. :(  I am attempting to use a 3rd party voip company for LD
   calling by doing the following:
  
   sip.conf
  
   [newtel]
   type=peer
   host=sip1.newtel.net
   nat=yes
   disallow=all
   allow=g729
  
   extentions.conf:
  
   [newtel]
   exten = _81.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
   exten = _81.,2,Congestion
 
  I think that your DIAL command should be:
 
  DIAL(SIP/newtel/${EXTEN:1})
 
   If someone can help me I will buy you a pony.
 
  medium-rare please ;-)
 
  Ryan
 
 
 
 No Pony for you :( same thing.
 
 Here is asterisk:
 
 Executing Dial(SIP/Phil-f9c7, SIP/newtel/16175554569) in new stack
 -- Called newtel/16175554569
  WARNING[16517]: chan_sip.c:687 retrans_pkt: Maximum retries exceeded
 on call [EMAIL PROTECTED] for seqno 102
 (Critical Request)
   == No one is available to answer at this time (1, 0/0/0)
 
 This is the packet from the firewall:
 
 16:56:32.665476 IP 172.17.2.10.sip 
 uslec-66-255-165-3.cust.uslec.net.sip: UDP, length: 703
 
 It just wont refer to the host.
 


No Pony for you :( same thing.

Here is asterisk:

Executing Dial(SIP/Phil-f9c7, SIP/newtel/16175554569) in new stack
   -- Called newtel/16175554569
WARNING[16517]: chan_sip.c:687 retrans_pkt: Maximum retries exceeded
on call [EMAIL PROTECTED] for seqno 102
(Critical Request)
 == No one is available to answer at this time (1, 0/0/0)

This is the packet from the firewall:

16:56:32.665476 IP 172.17.2.10.sip 
uslec-66-255-165-3.cust.uslec.net.sip: UDP, length: 703

It just wont refer to the host.

p.s. ryan. sorry for the dup messages.
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Re: [Asterisk-Users] Sip providor reference in extentions.conf

2005-01-06 Thread Philip Siegrist 
On Thu, 6 Jan 2005 18:34:03 -0500, Philip Siegrist [EMAIL PROTECTED] wrote:
 On Thu, 6 Jan 2005 18:30:40 -0500, Philip Siegrist [EMAIL PROTECTED] wrote:
  On Thu, 06 Jan 2005 16:13:38 -0700, Ryan Courtnage [EMAIL PROTECTED] 
  wrote:
   On Thu, 2005-06-01 at 17:56 -0500, Philip Siegrist wrote:
Hi,
   
I am stuck. :(  I am attempting to use a 3rd party voip company for LD
calling by doing the following:
   
sip.conf
   
[newtel]
type=peer
host=sip1.newtel.net
nat=yes
disallow=all
allow=g729
   
extentions.conf:
   
[newtel]
exten = _81.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _81.,2,Congestion
  
   I think that your DIAL command should be:
  
   DIAL(SIP/newtel/${EXTEN:1})
  
If someone can help me I will buy you a pony.
  
   medium-rare please ;-)
  
   Ryan
  
  
 
  No Pony for you :( same thing.
 
  Here is asterisk:
 
  Executing Dial(SIP/Phil-f9c7, SIP/newtel/16175554569) in new stack
  -- Called newtel/16175554569
   WARNING[16517]: chan_sip.c:687 retrans_pkt: Maximum retries exceeded
  on call [EMAIL PROTECTED] for seqno 102
  (Critical Request)
== No one is available to answer at this time (1, 0/0/0)
 
  This is the packet from the firewall:
 
  16:56:32.665476 IP 172.17.2.10.sip 
  uslec-66-255-165-3.cust.uslec.net.sip: UDP, length: 703
 
  It just wont refer to the host.
 
 
 No Pony for you :( same thing.
 
 Here is asterisk:
 
 Executing Dial(SIP/Phil-f9c7, SIP/newtel/16175554569) in new stack
-- Called newtel/16175554569
 WARNING[16517]: chan_sip.c:687 retrans_pkt: Maximum retries exceeded
 on call [EMAIL PROTECTED] for seqno 102
 (Critical Request)
  == No one is available to answer at this time (1, 0/0/0)
 
 This is the packet from the firewall:
 
 16:56:32.665476 IP 172.17.2.10.sip 
 uslec-66-255-165-3.cust.uslec.net.sip: UDP, length: 703
 
 It just wont refer to the host.
 
 p.s. ryan. sorry for the dup messages.
 

User error, Problem solved both syntaxt were correct. It was a dns issue. 

Ponies for all. Thanks
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