[asterisk-embedded] Problem with PSTN calls (Asterisk as SIP clienton embedded device)

Karl Schmutz kschmutz at starnetdata.com
Thu May 12 10:11:57 CDT 2011


A quick glance at your logs tells me that your SIP provider/endpoint at
66.8.50.218:5060 is rejecting the call sip:**********@sip.*****.co.za
SIP/2.0 <sip:**********@sip.*****.co.za%20SIP/2.0> (line 46) by saying:
"SIP/2.0 404 Not Found"

 

Although the number is filtered out for privacy reasons, I would double
check that the dialplan at endpoint at 66.8.50.218 will accept the
number referenced in sip: ********** @ sip.*****.co.za because that end
point is actively rejecting that number.

 

I haven't seen a post to this mailing list for a long time. You may get
better traction in the asterisk-users or in IRC as this seems to be more
related to standard SIP configuration.

 

 

Karl Schmutz
Networking Systems Engineer
Starnet Data Design, Inc.
Direct Line:  805.277.0117
Toll Free:  800.779.0587
Westlake Village, CA - Phoenix, AZ
www.starnetdata.com
Service. Value. Integrity.
Follow Us on Twitter:   www.twitter.com/Starnet_Data
<http://www.twitter.com/Starnet_Data>  

 

From: asterisk-embedded-bounces at lists.digium.com
[mailto:asterisk-embedded-bounces at lists.digium.com] On Behalf Of
helge.reikeras at gmail.com
Sent: Thursday, May 12, 2011 7:52 AM
To: asterisk-embedded at lists.digium.com
Subject: [asterisk-embedded] Problem with PSTN calls (Asterisk as SIP
clienton embedded device)

 

Hi 

I've spent two days trying to solve this issue but to no prevail and I'm
hoping to get some help.


I've configured Asterisk as a SIP client, running on OpenWRT on an
embedded device with onboard FXS and ATA. Asterisk is connecting to an
external SIP provider on the Internet who in turn provides a PSTN
gateway. I'm able to make calls to other SIP accounts registered on the
same server who are outside my LAN. However, I can not make calls to any
PSTN numbers. When trying to make PSTN calls it sounds like the person
at the other end is immediately rejecting the call although I know this
is not the case.

Firstly, I'm absolutely sure that the PSTN gateway is working because I
can make outbound PSTN calls with the same SIP account using other SIP
clients (Empathy-SIP, SIPDroid) from the same LAN. However, when
registering the same SIP account using Asterisk from OpenWRT all PSTN
calls fail. Inbound calls from PSTN numbers also fail while calls from
other SIP clients on the same server work fine. The SIP accounts shows
as registered in Asterisk.

 

I've attached detailed error logs. The log files 'messages-pstn.log'
shows the failed (PSTN) call and 'messages-voip.log' shows the
successful (VOIP) call. Note that I have replaced actual phone numbers
and domain names with *** for anonymity.

I suspect perhaps a codec issue, but I haven't been able to identify the
actual problem. Any ideas that will help me towards solving this problem
is greatly appreciated.

Regards,
Helge

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-embedded/attachments/20110512/eddf0a0b/attachment.htm>


More information about the asterisk-embedded mailing list