captshook: didn't you forget the payload?

Skinny is the Asterisk implementation of SCCP. The discussion did not consider the limitations of using SCCP (However a lot less work has gone into skinny than SIP and Asterisk generally cannot support direct media across different protocols, even if they both use RTP. Asterisk is basically a back to back user agent, with some optimisation when both parties are using the same protocol).

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308

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The discussion concludes that SCCP & SIP modules can run in parallel on Asterisk to support both SCCP & SIP devices. The same can also be done using Skinny but that has limited feature support for SCCP devices. To gain full access and class features - one should have sccp installed on the node. Please correct my statement and advise if the case if different. Thanks

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308

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Ok, Thanks for your answers.
I want to connect my sccp phones on an asterisk server and i want to create a sip trunk with 3CX PBX. In this case the sccp phones from Asterisk and sip phones from 3CX will communicate each other?

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308

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Yes SIP peers can talk to SCCP peers just like they can talk to IAX peers.

I have tried the SIP calling SCCP with Cisco phones.

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308

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Yes. Yes. (Although Asterisk will need to be a client for at least one of them.

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308

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Can i use a sip phone and a sccp phone on the same Asterisk server. Can these two devices communicate each other?

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308

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You don't need these, as there are other ways of doing this.

The sponsors of this forum would prefer that you used Digium branded equivalents if you are going the PCI route.

Statistics : Posted by arcath • on Thu Jun 12, 2014 6:10 am • Replies 1 • Views 55

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I have never really setup a Phone system before and I'm getting a bit stuck on the terminology but I think I've got an idea of what I need I'd just like an expert opinion.

We have 2 Phone lines coming into our office which are currently connected a PBX box which is a bit rubbish so we would like to swap to using Asterisk.

The thing I'm struggling with is getting the phone lines into the box I think I need this: http://www.voipon.co.uk/openvox-a400p02-p-670.html

Which I think will result in a system where we can have IP phones around the office on the Auto-Voice VLAN in our switches.

Also would I be able to add one of these (http://www.voipon.co.uk/openvox-fxs100-p-682.html) to that card letting us connect up our analogue FAX machine.

Is this right? or have I miss read everything?

Statistics : Posted by arcath • on Thu Jun 12, 2014 6:10 am • Replies 1 • Views 55

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I'm having a similar issue with Asterisk 12.3.2 and a Cisco 7970. Is the Cisco SIP stack not compatible with pjsip? Here's a copy of the pjsip trace and the asterisk console:

<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:25] NOTICE[26227]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678465/6709fec1da55176847f2a32dcef0dd82",opaque="172666d1538e1212",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


<--- Received SIP request (2069 bytes) from UDP:192.168.1.104:51657 --->
REFER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bKef4e619e
From: <sip:001380226e57@192.168.1.104>;tag=001380226e57000206b08360-bfeddbee
To: <sip:192.168.1.147>
Call-ID: 00138022-6e570002-e2def018-2282bbf6@192.168.1.104
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 1000 REFER
User-Agent: Cisco-CP7970G/9.3.1
Expires: 10
Max-Forwards: 70
Contact: <sip:001380226e57@192.168.1.104:5060>
Require: norefersub
Referred-By: <sip:001380226e57@192.168.1.104>
Refer-To: cid:dbbc3f90@192.168.1.104
Content-Id: <dbbc3f90@192.168.1.104>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 1308
Content-Type: application/x-cisco-alarm+xml
Content-Disposition: session;handling=required

<?xml version="1.0" encoding="UTF-8"?>
<x-cisco-alarm>
<Alarm Name="LastOutOfServiceInformation">
<ParameterList>
<String name="DeviceName">SEP001380226E57</String>
<String name="DeviceIPv4Address">192.168.1.105/24</String>
<String name="IPv4DefaultGateway">192.168.1.1</String>
<String name="DeviceIPv6Address"></String>
<String name="IPv6DefaultGateway"></String>
<String name="ModelNumber">CP-7970G</String>
<String name="NeighborIPv4Address"></String>
<String name="NeighborIPv6Address"></String>
<String name="NeighborDeviceID"></String>
<String name="NeighborPortID"></String>
<Enum name="DHCPv4Status">1</Enum>
<Enum name="DHCPv6Status">0</Enum>
<Enum name="TFTPCfgStatus">1</Enum>
<Enum name="DNSStatusUnifiedCM1">4</Enum>
<Enum name="DNSStatusUnifiedCM2">0</Enum>
<Enum name="DNSStatusUnifiedCM3">0</Enum>
<String name="VoiceVLAN">4095</String>
<String name="UnifiedCMIPAddress">192.168.1.147</String>
<String name="LocalPort">0</String>
<String name="TimeStamp">13600306518511360028803907</String>
<Enum name="ReasonForOutOfService">14</Enum>
<String name="LastProtocolEventSent">Sent:REGISTER sip:192.168.1.147 SIP/2.0 Cseq:101 REGISTER CallId:00138022-6e570005-4677c4c9-d2468146@192.168.1.104</String>
<String name="LastProtocolEventReceived"></String>
</ParameterList>
</Alarm>
</x-cisco-alarm>
[2014-06-13 11:54:26] NOTICE[27538]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:001380226e57@192.168.1.104>' failed for '192.168.1.104:51657' (callid: 00138022-6e570002-e2def018-2282bbf6@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (521 bytes) to UDP:192.168.1.104:51657 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bKef4e619e
Call-ID: 00138022-6e570002-e2def018-2282bbf6@192.168.1.104
From: <sip:001380226e57@192.168.1.104>;tag=001380226e57000206b08360-bfeddbee
To: <sip:192.168.1.147>;tag=z9hG4bKef4e619e
CSeq: 1000 REFER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678466/1f0291f85a02413bdb80fff02ebee0d8",opaque="575bd04d74452b64",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:26] NOTICE[26227]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678466/adc0052802d0ee690398bbe51fb2c29b",opaque="7f40e9672738bcce",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


[2014-06-13 11:54:27] ERROR[26232]: pjsip:0 <?>: sip_transport. Error processing 2069 bytes packet from UDP 192.168.1.104:51412 : PJSIP syntax error exception when parsing '' header on line 1 col 1:
<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:27] NOTICE[27538]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678467/fccc661771763b7b1d601dbe5d848c51",opaque="75f3a88924d1dbd2",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:29] NOTICE[26227]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678469/2ee3c94bc8785a1ed82208dd941cea1c",opaque="52e5a0a52a43532b",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:33] NOTICE[27538]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678473/999fb1a94a416c2e2f64f18edb6ab304",opaque="5e41b9d0576492ef",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:37] NOTICE[26227]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678477/213a5011a82bb6e5495d99c139248f29",opaque="2b9efa2d5d05e60f",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:41] NOTICE[27538]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678481/5061ad69d57bd2192dcbd3d38fad0661",opaque="218630de299eb2cb",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0


<--- Received SIP request (958 bytes) from UDP:192.168.1.104:52767 --->
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK516ff916
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
Max-Forwards: 70
Date: Tue, 05 Feb 2013 02:24:46 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/9.3.1
Contact: <sip:1002@192.168.1.104:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001380226e57>";+u.sip!devicename.ccm.cisco.com="SEP001380226E57";+u.sip!model.ccm.cisco.com="30006"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:14 Name=SEP001380226E57 Load=SIP70.9-3-1SR2-1S Last=cm-closed-tcp"
Expires: 3600


[2014-06-13 11:54:45] NOTICE[26227]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.104:52767 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;received=192.168.1.104;branch=z9hG4bK516ff916
Call-ID: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104
From: <sip:1002@192.168.1.147>;tag=001380226e5700034fdb05c0-6209740e
To: <sip:1002@192.168.1.147>;tag=z9hG4bK516ff916
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1402678485/7b372e66a78c0acbeaf60f0c84ced233",opaque="26e7eaf91d146ade",algorithm=md5,qop="auth"
Server: FPBX-12.0.1alpha51(12.3.2)
Content-Length: 0

[2014-06-13 11:54:57] NOTICE[27538] res_pjsip/pjsip_distributor.c: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:52767' (callid: 00138022-6e570002-bcbd9548-1c08fb66@192.168.1.104) - No matching endpoint found
[2014-06-13 11:55:13] NOTICE[26227] res_pjsip/pjsip_distributor.c: Request from '<sip:001380226e57@192.168.1.104>' failed for '192.168.1.104:52243' (callid: 00138022-6e570003-9aa4af96-77801626@192.168.1.104) - No matching endpoint found
[2014-06-13 11:55:13] NOTICE[27538] res_pjsip/pjsip_distributor.c: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:50964' (callid: 00138022-6e570003-d5879e16-86033da6@192.168.1.104) - No matching endpoint found
[2014-06-13 11:55:14] NOTICE[26227] res_pjsip/pjsip_distributor.c: Request from '<sip:001380226e57@192.168.1.104>' failed for '192.168.1.104:49271' (callid: 00138022-6e570003-9aa4af96-77801626@192.168.1.104) - No matching endpoint found
[2014-06-13 11:55:14] NOTICE[27538] res_pjsip/pjsip_distributor.c: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:50964' (callid: 00138022-6e570003-d5879e16-86033da6@192.168.1.104) - No matching endpoint found
[2014-06-13 11:55:15] NOTICE[26227] res_pjsip/pjsip_distributor.c: Request from '<sip:1002@192.168.1.147>' failed for '192.168.1.104:50964' (callid: 00138022-6e570003-d5879e16-86033da6@192.168.1.104) - No matching endpoint found

Statistics : Posted by krunner • on Fri Mar 28, 2014 12:53 pm • Replies 4 • Views 413

---

That REGISTER sequence is incomplete, your phone must re send the REGISTER again with the www digest auth in it in order to register.

Try capturing the complete log, if thats all, then your phone is not responding the challenge.

Statistics : Posted by krunner • on Fri Mar 28, 2014 12:53 pm • Replies 4 • Views 413

---

After running debug this is what I got:

<--- Received SIP request (683 bytes) from UDP:172.X.X.X:49162 --->
REGISTER sip:172.X.X.X SIP/2.0
Via: SIP/2.0/UDP 172.X.X.X:5060;branch=z9hG4bKe6bf3a86
From: <sip:10405@172.X.X.X>;tag=0023049a8xxxxxxxxf434118-2d3065ce
To: <sip:10405@172.X.X.X>
Call-ID: 0023049a-xxxxxxxx-16c11800-38b70df6@172.X.X.X
Max-Forwards: 70
Date: Tue, 05 May 2009 20:36:25 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7941G/8.5.2
Contact: <sip:10405@172.X.X.X:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-002304xxxxxx>";+u.sip!model.ccm.cisco.com="115"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP002304XXXXXX Load=SIP41.8-5-2S Last=initialized"
Expires: 3600


<--- Transmitting SIP response (478 bytes) to UDP:172.X.X.X:49162 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.X.X.X:5060;rport;received=172.X.X.X;branch=z9hG4bKe6bf3a86
Call-ID: 0023049a-xxxxxxxx-16c11800-38b70df6@172.X.X.X
From: <sip:10405@172.X.X.X>;tag=0023049xxxxxxxx23f434118-2d3065ce
To: <sip:10405@172.X.X.X>;tag=z9hG4bKe6bf3a86
CSeq: 101 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1396042718/1243b24bxxxxxxxx67f4a714789d6f24",opaque="294e3470xxxxxxxx",algorithm=md5,qop="auth"
Content-Length: 0

Statistics : Posted by krunner • on Fri Mar 28, 2014 12:53 pm • Replies 4 • Views 413

---

Enable pjsip debug.

Statistics : Posted by krunner • on Fri Mar 28, 2014 12:53 pm • Replies 4 • Views 413

---

I installed Asterisk 12.1.1 from source on CentOS 6.5 and initially configured it to work with SIP. I was able to get all devices working including X-lite, a Polycom vvx1500 and the Cisco 7941. Everything worked fine including video.

I recompiled Asterisk without chan_sip to get it working with only pjsip. I have since been able to get X-lite to X-lite audio working, the Polycom vvx1500 audio. The Cisco 7941 however is stuck registering. I used the following to configure pjsip.conf

;===============TRANSPORT

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

;===============ENDPOINT TEMPLATES

[endpoint-basic](!)
type=endpoint
transport=simpletrans
context=internal
disallow=all
allow=ulaw

[auth-userpass](!)
type=auth
auth_type=userpass

[aor-single-reg](!)
type=aor
max_contacts=1

;===============EXTENSION 6001

[6001](endpoint-basic)
auth=auth6001
aors=6001

[auth6001](auth-userpass)
password=6001
username=6001

[6001](aor-single-reg)

Any suggestions?

Statistics : Posted by krunner • on Fri Mar 28, 2014 12:53 pm • Replies 4 • Views 413

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I did not have the latest version. I downloaded the latest version and reinstalled. It is now working.

Statistics : Posted by tomrgsd • on Fri Jun 13, 2014 9:37 am • Replies 1 • Views 61

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I have installed Asterisk, and configured the network interface, but cannot communicate on the network. If I start pinging the machine while booting, it will respond to ping for about 6 responses before it gets to a login prompt. Once at login, it stops responding and I cannot ping out from the box either. I thought maybe the firewall was the issue, but I stop the service and it still will not communicate. I have reinstalled twice with the same results.

Statistics : Posted by tomrgsd • on Fri Jun 13, 2014 9:37 am • Replies 1 • Views 61

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Just as an update I am also getting the same result when submitting the request via XML. I have opened a ticket with Digium to take a look at the issue and will update with its progress.

Statistics : Posted by linkn3echo • on Tue Apr 29, 2014 2:06 pm • Replies 1 • Views 240

---

Hello,

We currently use Switchvox:

Code:

Current Version:
5.8.3.1 (61925)
Maximum User Extensions: 70 (currently using 70)
Phone Feature Packs: 10 (currently using 0)
Subscription Expiration:October 3 2017

I am trying to use the API to make a full backup, i.e including all directories available, however it appears that specifying the directory id and size is not being taken into account for the backup.

A full backup from the web interface, usually consists of 400MB

Code:

4/28/2014 3:56 PM
399.7 MB
Voicemail and Fax Files
Voicemail Greetings
Music On Hold Files
PBX Error Logs

4/25/2014 6:33 PM
399.3 MB
Voicemail and Fax Files
Voicemail Greetings
Music On Hold Files
PBX Error Logs

4/20/2014 11:32 AM
393.9 MB
Voicemail and Fax Files
Voicemail Greetings
Music On Hold Files
PBX Error Logs

Running the following command creates a backup without any of the optional directories:

Request

Code:

curl -k -H "Content-Type: application/json" -X POST -d '{"request": {"method": "switchvox.backups.add" }}' --digest -u pbx_user:pbx_password https://pbx.somecompany.net/json

Response

Code:

{
   "response" : {
      "method" : "switchvox.backups.add",
      "result" : {
         "progress" : {
            "id" : "dukjvzznkedhwexs"
         }
      }
   }
}

Progress

Code:

curl -k -H "Content-Type: application/json" -X POST -d '{"request": {"method": "switchvox.progress.check","parameters": { "progress_id": "dukjvzznkedhwexs" }}}' --digest -u pbx_user:pbx_password https://pbx.somecompany.net/json

Code:

                                                                                                       
{
   "response" : {
      "method" : "switchvox.progress.check",
      "result" : {
         "progress" : {
            "status" : "OK",
            "percentage_done" : "",
            "id" : "dukjvzznkedhwexs",
            "message" : "Backup complete",
            "state" : "done",
            "backup_id" : "65"
         }
      }
   }
}

Querying data for the backup id of 65 you get:
65 | 2014-04-29 | 15:16:40.003279 | 108594663KB |

Now running the curl command including the directories that I want to include does not seem to change the size of the backup at all.

Below is the directory name, size (KB), and ID. I want the backup to include directories 1,15,2,and 3.

Code:

Directory Name                 Directory Size       Directory ID       
---------------                --------------       ------------       
PBX Configuration              135737               5                   
Hardware Configuration         4228                 7                   
Database Configuration         147543               9                   
Sounds / Sound Packs           158322500            6                   
Profile Images                 470132               16                 
IAX RSA Keys                   6619                 12                 
Audio Codecs                   4421                 13                 
Distinctive Ringtones          12160062             17                 
Idle Screens                   4096                 18                 
Phone apps                     4096                 19                 
Voicemail and Fax Files        520050949            1                   
Voicemail Greetings            6825761              15                 
Music On Hold Files            22979525             2                   
PBX Error Logs                 180345994            3   

Code:

curl -k -H "Content-Type: application/json" -X POST -d '{"request": {"method": "switchvox.backups.add", "parameters": {"directories": [ "1" , "520050949", "15", "6825761", "2", "22979525", "3", "180345994" ] }}}' --digest -u pbx_user:pbx_password https://pbx.somecompany.net/json 

Response

Code:

{
   "response" : {
      "method" : "switchvox.backups.add",
      "result" : {
         "progress" : {
            "id" : "wpyayfsqmnyetcba"
         }
      }
   }
}

Progress

Code:

{
   "response" : {
      "method" : "switchvox.progress.check",
      "result" : {
         "progress" : {
            "status" : "OK",
            "percentage_done" : "",
            "id" : "wpyayfsqmnyetcba",
            "message" : "Backup complete",
            "state" : "done",
            "backup_id" : "66"
         }
      }
   }
}

Querying data for the backup id of 66 you get:
Backup Result

Code:

| 66 | 2014-04-29 | 15:39:12.760367 | 108611802KB |

I also tried:

Code:

curl -k -H "Content-Type: application/json" -X POST -d '{"request": {"method": "switchvox.backups.add", "parameters": {"directories": { "directory": [ "1" , "519199216", "15", "6825761", "2", "22979525", "3", "179945481" ]}}}}' --digest -u pbx_user:pbx_password https://pbx.somecompany.net/json

Response

Code:

{
   "response" : {
      "method" : "switchvox.backups.add",
      "result" : {
         "progress" : {
            "id" : "bkwwnirqzlovktff"
         }
      }
   }
}

Backup Result

Code:

67 2014-04-29 15:49:49.549255 108599401

I tried using the test suite but it gives methods and options that do not exist in the documentation. There also seems to be some difference on the example of how an XML request and JSON request.

http://developers.digium.com/switchvox/ ... st_Formats

If I plug the XML into a JSON converter at http://dfsq.github.io/xml2json/#/xml/paste and look at the JSON output I get:

Code:

{"request":{"method":"switchvox.extensions.getInfo","parameters":{"extensions":{"extension":["100","101"]}}}}

which is different from the example of:

Code:

{"request": {"method": "switchvox.extensions.getInfo", "parameters": {"extensions": [ "100",  "101" ]}}}

If I use the digium test suite, the available method catagory and method are for version 4.6 and don't include options available in the current API documentation. For example this is the output of the request section when choosing, JSON, Backups, switchvox.backups.add.

Code:

{"request": {"method": "switchvox.backups.add", "parameters": {"optional_directory_ids": [], "progress_id": ""}}}

http://developers.digium.com/switchvox/ ... =testSuite

Statistics : Posted by linkn3echo • on Tue Apr 29, 2014 2:06 pm • Replies 1 • Views 240

---

Hello All,

- Asterisk 11.x , fresh install

Been using asterisk for about 5 years now,and always learning. My new Asterisk 11 setup on CentOS6 installed via rpms from asterisk/digium repos.

This is a two nic server that is also a linux terminal server,just for completeness.
eth0= internal interface
eth1=external interface

I have been tryin to learn some new things in regards to QoS and the miriad of traffic shaping docs that are out there.

In doing a lot of of tcpdump,and Wireshark captures ,come to find out, that when traffic is passed from eth0,to eth1 looking at Wireshark the udp 'port(s) shows from sip(eth0) > STUN(eth1) and RTP(eth0), to > CLASSIC(eth1).

I am thinking due to tthis situation all of the classic Qos/traffic shaping scripts that are out there is not,working,,due to the fact the packets being passed are being tagged differently of being the standrard SIP/RTP,by the time they go out eth1(external interface)..
The voice quality is pretty good,but from time to time, i do get some voice breakup,on the receiver's end. I can always hear who i am talking to fine.

I have tried using both NAT and No Nat in the sip settings(I am using FreePBX GUI),in conjunction with asterisk. They both work fine but makes no difference in regards to the udp packets being tagged diffrently when being passed from eth0, to eth1.

Also for completeness I have only tried using my Android Google Voice account.Could this be part of the situation,I am seeing here.

Has anyone experienced this situation,and if so,how do i setup any of the ,,,,example sipshaper scritps to work?

Thank You,
Barry

Statistics : Posted by brcisna • on Fri Jun 13, 2014 5:32 pm • Replies 0 • Views 18

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NOTICEs are not error messages, although in this case they are reporting errors.

Bad FCS means that there is a frame check sequence error, i.e. a block of data has been received which fails its self check.

Abort means that more consecutive ones have been received than are allowed (there needs to be a zero, which is removed from the data, after every 5 (I think) ones).

Both of these indicate either very poor signal quality or incompatible settings between the two ends of the link.

As this is a Digium card, you should use Digium's commercial support service, not this open source peer support forum.

For that, you will need the serial number of the card.

More generally, you will need to identify what is on the other end of the line, the name of service product, if it is a standard service, or details of how it is configured, otherwise, and also how it has been configured for both Asterisk and Dahdi at your end.

Generally, the PSTN should provide timing, so if this is a PSTN link, check that you are letting it do that.

Statistics : Posted by jiteshright • on Sat Jun 14, 2014 12:37 am • Replies 1 • Views 90

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[Jun 14 12:01:47] NOTICE[4204]: chan_dahdi.c:10518 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
[Jun 14 12:01:47] NOTICE[4204]: chan_dahdi.c:10518 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1

repeat and repeat...

Statistics : Posted by jiteshright • on Sat Jun 14, 2014 12:37 am • Replies 1 • Views 90

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