[asterisk-users] Calling issue for non-extension numbers

[Danny Nicholas]

It is probably a dialplan or timeout issue.  What happens if you do
80055511212# ?

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kayton Sapale
Sent: Tuesday, August 04, 2009 4:13 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Calling issue for non-extension numbers

 

Hi all,

Thanks to the previous replies that helped me with this before, but I got
side-tracked in the middle of trying to figure this out, so apologies for
posting the same issue.  I use a Nokia e71, with an asterisk server and am
having an issue dialing certain numbers.  When I attempt to dial a local
number, like xxx-xxx-xxxx, I cannot connect.  What shows in the asterisk
debug is the following:

Found peer '104'

However, if I try to dial an extension that is configured on the asterisk
server, the call goes through fine.  When I use another device to connect
the server (another nokia actually) and dial a local number like
xxx-xxx-xxxx, I see this in the debug dialog:

Found peer '103' Found RTP audio format 96 Found RTP audio format 0 Found
RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found
RTP audio format 98 Found RTP audio format 13 Peer audio RTP is at port
192.168.111.183:49152 Found unknown media description format AMR for ID 96
Found audio description format PCMU for ID 0 Found audio description format
PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio
description format G729 for ID 18 Found audio description format
telephone-event for ID 98 Found audio description format CN for ID 13
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x50c
(ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3
(telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at
port 192.168.111.183:49152 Looking for 6789940793 in DLPN_Free_Outbound
(domain sip.speartek.com) list_route: hop:  <mailto:sip:103 at 192.168.111.183>
<sip:103 at 192.168.111.183>

It appears that my device cannot connect to the server when dialing certain
numbers.  Does anyone have any idea about this?

Thanks,
Kayton 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090804/1462ab4f/attachment.htm