From helge.reikeras at gmail.com Thu May 12 09:52:13 2011 From: helge.reikeras at gmail.com (helge.reikeras at gmail.com) Date: Thu, 12 May 2011 16:52:13 +0200 Subject: [asterisk-embedded] Problem with PSTN calls (Asterisk as SIP client on embedded device) Message-ID: 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... 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Name: messages-voip.log Type: text/x-log Size: 11437 bytes Desc: not available URL: From kschmutz at starnetdata.com Thu May 12 10:11:57 2011 From: kschmutz at starnetdata.com (Karl Schmutz) Date: Thu, 12 May 2011 08:11:57 -0700 Subject: [asterisk-embedded] Problem with PSTN calls (Asterisk as SIP clienton embedded device) In-Reply-To: References: Message-ID: 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 (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 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: From helge.reikeras at gmail.com Thu May 12 10:48:17 2011 From: helge.reikeras at gmail.com (helge.reikeras at gmail.com) Date: Thu, 12 May 2011 17:48:17 +0200 Subject: [asterisk-embedded] Problem with PSTN calls (Asterisk as SIP clienton embedded device) In-Reply-To: References: Message-ID: Thanks Karl. I'll give the user list a try. Helge On May 12, 2011 5:14 PM, "Karl Schmutz" wrote: > 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 (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 > > > > > 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: