[asterisk-users] SIP Woes

[tim robinson]

Hi Dave
The problem is with the way in which Asterisk handles 'overlap' dialling 
with SIP.  i.e. not very well at all.  If you remove the early dial 
feature from the phone I think you will find it will solve the problem.

The issue is that Asterisk does not apply the digit timeout on SIP early 
dial. To behave like chan_zap does with overlap dialling  the timeout 
MUST be in Asterisk.

 I have had a long email exchange with Olle, master of chan_sip but he 
told me that SIP was never  designed to operate in this manner for 
overlap dialling.  If so, I think this is actually a major flaw in 
SIP...Unfortunately multi-length dialplans are a fact of life in many 
countries, which makes SIP non-optimal.

Rgds
Tim Robinson
Basingstoke, UK



Dave Hope wrote:
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>
> Hello all,
>
> I've been trying to play with asterisk (after a two month break) and am
> having some problems getting my SIP connection to a third party provider
> to work. In the asterisk console I notice:
>
> - -------------------------
> debian*CLI> set verbose 999
> Verbosity was 0 and is now 999
> Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:2355 sip_alloc: Allocating new
> SIP call for 88e1fcfa-c618-db11-91fa-000fea3f84d4 at localhost
> Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting
> NAT on RTP to 4
> Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping
> retransmission on '88e1fcfa-c618-db11-91fa-000fea3f84d4 at localhost' of
> Response 1: Found
> Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting
> NAT on RTP to 4
> Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:7329 handle_request: Check for
> res for 200
> Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:1620 update_user_counter: Call
> from user '200' is 1 out of 0
> Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping
> retransmission on '88e1fcfa-c618-db11-91fa-000fea3f84d4 at localhost' of
> Response 2: Found
> - -------------------------
>
> I believe that's some sort of SIP routing issue related to ReInvite's ?
> - - Is there a workaround for this? In the attempt that someone may be
> able to shed some light on the matter, I've uploaded my current
> configuration to:
>
> 	http://files.davehope.co.uk/asterisk-problem/
>
> I've also uploaded the output of 'sip debug'. The interesting bit in
> that (to me at least) is the message:
>
> - -------------------------
> Looking for 100000 in Outgoing
> Reliably Transmitting (NAT):
> SIP/2.0 484 Address Incomplete
> Via: SIP/2.0/UDP
> - -------------------------
>
> Is it so simple that I've missed something out in my outgoing bit on my
> dialplan ? Anyway, the complete log can be found here:
>
> 	http://files.davehope.co.uk/asterisk-problem/debug.log
>
> Ohh. And:
>
> - -------------------------
> root at debian:/etc/asterisk# asterisk -V
> Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k
> - -------------------------
>
> If anyone would be so kind as to shed some insight into the matter it'd
> be greatly appreciated!,
>
> Kind Regards,
>
> Dave
>
>
>
>
> 	
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