Hi,
I have an asterisk-13.1.1 config on fedora22 that is not working after an upgrade from fedora20.

I've compiled and installed dahdi-linux from git and now the dahdi modules don't install on boot. If I install them and a few others manually, asterisk still doesn't seem to like it:

Code:

# lsmod|egrep 'dahdi|wctdm'
dahdi_echocan_mg2      16384  0
wctdm                  49152  0
dahdi                 229376  2 wctdm,dahdi_echocan_mg2
crc_ccitt              16384  2 dahdi,isdnhdlc

Code:

# asterisk -f
XSLT support not found. XML documentation may be incomplete.
[Jul 18 15:04:06] NOTICE[8731]: cdr.c:4156 cdr_toggle_runtime_options: CDR simple logging enabled.
[Jul 18 15:04:07] NOTICE[8731]: loader.c:1323 load_modules: 221 modules will be loaded.
[Jul 18 15:04:08] WARNING[8731]: res_phoneprov.c:1229 get_defaults: Unable to find a valid server address or name.
[Jul 18 15:04:08] ERROR[8731]: ari/config.c:296 process_config: No configured users for ARI
[Jul 18 15:04:08] WARNING[8731]: loader.c:564 load_dynamic_module: Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Jul 18 15:04:08] WARNING[8731]: loader.c:1038 load_resource: Module 'res_ari_mailboxes.so' could not be loaded.
[Jul 18 15:04:09] WARNING[8731]: res_musiconhold.c:1944 load_module: No music on hold classes configured, disabling music on hold.
15:04:09.799 os_core_unix.c !pjlib 2.3 for POSIX initialized
15:04:09.800          pjlib !select() I/O Queue created (0x7fdfdc0008e8)
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:4077 dahdi_open: Unable to specify channel 1: Invalid argument
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:12013 mkintf: Unable to open channel 1: Invalid argument
here = 0, tmp->channel = 1, channel = 1
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:17420 build_channels: Unable to register channel '1'
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:17627 process_dahdi: Channel '1' failure ignored: ignore_failed_channels.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:4077 dahdi_open: Unable to specify channel 2: Invalid argument
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:12013 mkintf: Unable to open channel 2: Invalid argument
here = 0, tmp->channel = 2, channel = 2
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:17420 build_channels: Unable to register channel '2'
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:17627 process_dahdi: Channel '2' failure ignored: ignore_failed_channels.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:4077 dahdi_open: Unable to specify channel 3: Invalid argument
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:12013 mkintf: Unable to open channel 3: Invalid argument
here = 0, tmp->channel = 3, channel = 3
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:17420 build_channels: Unable to register channel '3'
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:17627 process_dahdi: Channel '3' failure ignored: ignore_failed_channels.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:4077 dahdi_open: Unable to specify channel 4: Invalid argument
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:12013 mkintf: Unable to open channel 4: Invalid argument
here = 0, tmp->channel = 4, channel = 4
[Jul 18 15:04:09] ERROR[8731]: chan_dahdi.c:17420 build_channels: Unable to register channel '4'
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:17627 process_dahdi: Channel '4' failure ignored: ignore_failed_channels.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:18925 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:18925 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:18925 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:18925 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Jul 18 15:04:09] WARNING[8731]: chan_dahdi.c:18925 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Jul 18 15:04:09] NOTICE[8781]: chan_sip.c:23798 handle_response_peerpoke: Peer '7003' is now Reachable. (15ms / 2000ms)
[Jul 18 15:04:12] NOTICE[8731]: confbridge/conf_config_parser.c:2047 verify_default_profiles: Adding default_menu menu to app_confbridge
[Jul 18 15:04:12] NOTICE[8731]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jul 18 15:04:12] WARNING[8731]: pbx.c:9790 add_priority: Unable to register extension '_X' priority 1 in 'mainmenu-night', already in use
[Jul 18 15:04:12] WARNING[8731]: pbx_config.c:1850 pbx_load_config: Unable to register extension at line 220 of extensions.conf
[Jul 18 15:04:12] WARNING[8731]: pbx.c:9790 add_priority: Unable to register extension 's' priority 2 in 'incoming-sales', already in use
[Jul 18 15:04:12] WARNING[8731]: pbx_config.c:1850 pbx_load_config: Unable to register extension at line 291 of extensions.conf
[Jul 18 15:04:12] ERROR[8731]: codec_dahdi.c:820 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory
[Jul 18 15:04:12] WARNING[8731]: app_voicemail.c:13548 actual_load_config: maxsilence should be less than minsecs or you may get empty messages

I'm trying to understand why the dahdi modules don't automatically get loaded on boot. How is that controlled? It appears fedora22 has /etc/init.d/dahdi that runs "systemctl start dahdi.service", and while it reports that it ran, it never loads the modules.

/etc/modules.d/dahdi.conf is all comments, and /etc/modules.d/dahdi.blacklist.conf has my driver blacklisted:

Code:

blacklist wctdm

I really have no idea what to do next and have been at this for more than five hours.

Statistics : Posted by gossamer • on Sat Jul 18, 2015 1:08 pm • Replies 0 • Views 57

---

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 12 • Views 3431

---

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 12 • Views 3431

---

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 12 • Views 3431

---

Enable pjsip debug.

Statistics : Posted by krunner • on Fri Mar 28, 2014 12:53 pm • Replies 12 • Views 3431

---

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 12 • Views 3431

---

Hi David,

Please suggest me the solution regarding this.See, I have actually two doubts:
1) Does Digium PRI card (TE4200) supports INAP
2) If yes, Where do I find the suitable library.

If this is not the right place to ask this question then where should I post this doubt . Please answer accordingly.

Statistics : Posted by vivek_raj • on Fri Jul 17, 2015 12:28 pm • Replies 3 • Views 232

---

That sounds more like a developer question.

Statistics : Posted by vivek_raj • on Fri Jul 17, 2015 12:28 pm • Replies 3 • Views 232

---

Thanks for your reply !
Could you please tell me that whether there is any library of SS7 which supports INAP protocol.

Statistics : Posted by vivek_raj • on Fri Jul 17, 2015 12:28 pm • Replies 3 • Views 232

---

I suspect you have to purchase software that supports it.

For the hardware question, you should ask Digium the company, not the Asterisk opens source community.

However, if it is built on SS7, I imagine that it will differ at a level where the hardware is not involved.

Statistics : Posted by vivek_raj • on Fri Jul 17, 2015 12:28 pm • Replies 3 • Views 232

---

I am currently working on Missed Call project in India. I have setup the whole project on Asterisk (using CentOS 6.6 as a base OS).We have deployed this project with different telecom provider on SIP, SS7 etc.

One of the telecom provider is providing us the connectivity on SS7-TDM using INAP protocol.They have provided us the GT and SPC . Now the challenge is we don't have any library which supports INAP. I have a few doubts which is listed below:
1) Does Digium card(TE420) supports INAP
2) Do We have to purchase some other hardware device to handle this INAP protocol.

Currently, we are getting following logs for the same configuration :

[1] Len = 196 [ 92 fe 3f c3 c0 0b cc 58 09 81 03 0e 19 0b 12 0c 00 12 04 19 87 63 39 99 81 0b 12 0c 00 12 04 19 89 19 30 00 01 9e 62 81
9b 48 04 5e c8 97 e2 6b 1e 28 1c 06 07 00 11 86 05 01 01 01 a0 11 60 0f 80 02 07 80 a1 09 06 07 04 00 01 01 01 00 00 6c 73 a1 71 02 01
00 02 01 00 30 69 80 02 03 78 82 06 01 10 bb 47 99 f2 83 07 03 11 99 11 00 35 84 85 01 0a 89 01 01 8a 08 04 93 19 89 19 30 00 60 af 30
30 0a 02 01 15 a1 05 04 03 00 00 08 30 0f 02 01 17 a1 0a 30 08 04 04 98 91 00 ee 05 00 30 08 02 01 31 a1 03 83 01 11 30 07 02 01 3b a1
02 30 00 97 02 91 81 9a 02 60 01 bb 05 80 03 80 90 a3 9c 01 03 ]
[1] FSN: 126 FIB 1
[1] BSN: 18 BIB 1
[1] <[0] MSU
[1] [ 92 fe 3f ]
[1] Network Indicator: 3 Priority: 0 User Part: SCCP (3)
[1] [ c3 ]
[1] OPC 9008 DPC 3008 SLS 5
[1] [ c0 0b cc 58 ]
[1]
[1] Unable to process message destined for userpart 3; dropping message


I hope, I have described you in details

Statistics : Posted by vivek_raj • on Fri Jul 17, 2015 12:28 pm • Replies 3 • Views 232

---

I am interested in setting up an on-site PBX with Asterisk. I'm new to Asterisk, and I'm still trying to figure out which version of Asterisk and associated GUI to use. Is AsteriskNOW the best choice? What are any limitations that AsteriskNOW has? Do you have to buy things to run a PBX successfully with AsteriskNOW?

Thanks for your help.

Statistics : Posted by kchen16 • on Mon Jul 20, 2015 11:24 am • Replies 1 • Views 61

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ETA is by the end of the year, I believe.

if you've got a client that needs some of the enhancements now, it's available on DCS (Switchvox Cloud) now.

Statistics : Posted by lprikockis • on Mon Jul 20, 2015 8:36 am • Replies 1 • Views 81

---

Has anyone heard any information, rumors, rumblings etc. as to when Switchvox 6.0 might be coming to appliances?

Statistics : Posted by lprikockis • on Mon Jul 20, 2015 8:36 am • Replies 1 • Views 81

---

Hi,

I just thought I'd give this thread a bump.

Statistics : Posted by intellectit • on Mon Jun 22, 2015 4:51 am • Replies 1 • Views 687

---

Hi,

Does anyone know of a method to retrieve the first six digits of the registration code? I'm pretty sure that there used to be such a method in an earlier version of the API. However I can't see tha tit exists any more in the current documentation.

Many thanks.

Statistics : Posted by intellectit • on Mon Jun 22, 2015 4:51 am • Replies 1 • Views 687

---

1) Use the subroutine option of the dial application.

2) Use ISDN rather than analogue, or get a line with answer supervision and configure Asterisk to use it. Asterisk supports line reversal answer supervision. Enabling it in Asterisk when the network operator hasn't enabled it on the line will result in Asterisk assuming the call has never been answered. Any non-trivial use of the PSTN should always interface vis ISDN; that is is how ITSPs interface. Analogue lines often don't have any supervision and those with disconnect supervision cannot distinguish between CLEAR and RELEASE, so you can have to wait for the network RELEASE timeout before an outgoing call clears.

Statistics : Posted by loidangthanh • on Mon Jul 20, 2015 9:30 pm • Replies 1 • Views 42

---

Hi guys, I'm newsterisk, I have the following task, and i don't know how to solve it.

I have a context with its own default extension

Code:

[my-context]
exten => s,1,Answer()
        same => n,Dial(SIP/817,10,tTwWA(demo-congrats))
        same => n,Hangup()

I run it with originate command in CLI, or in AMI via telnet.

Code:

originate SIP/814 extension s@my-context

It's simple and works in sequence

1. SIP/814 ringing
2. I answer SIP/814
3. SIP/817 ringing
4. Another guy answers SIP/817
5. Playing announcement at SIP/817
6. Hangup

But it does not match my requirement yet, I don't want the calling party (SIP/814 in this case) to join the sequence, I just want to give the called party (SIP/817) the announcement, and I don't know how to do that stuff.

Btw, when called party is from the DAHDI source (eg DAHDI/1/${my-cellphone-number}), the announcement played before I accept the call, so I just can hear end part of the announcement. How would I fix that?

Any help would be appreciated.
Best Regard
Loi Dang

Statistics : Posted by loidangthanh • on Mon Jul 20, 2015 9:30 pm • Replies 1 • Views 42

---

No.

Even if there was, 180 Ringing doesn't mean the destination just started to emit a sound.

Statistics : Posted by telemagic • on Mon Jul 20, 2015 5:12 pm • Replies 1 • Views 59

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Hi Guys

I am wondering if there is any way to set a timeout on Dial command to run from when Asterisk recieves the 180 Ringing message from the SIP channel instead of from when the call is placed?

If anyone could point me in the right direction of how to do this I would appreciate it!

Regards

Statistics : Posted by telemagic • on Mon Jul 20, 2015 5:12 pm • Replies 1 • Views 59

---