Cisco 7975 connecting via PJSIP

needmorebeerformewal · June 29, 2022, 12:42pm

I have a Cisco 7975 that just keeps says registering. If i use the same account details using a software client this connects fine and i can see it in the log.

When trying to connect using the phone i get the following and the phone just keeps saying registering

[2022-06-29 07:20:46] SECURITY[2514] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2022-06-29T07:20:46.654+0000",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="302",SessionID="e80462eb-18ef0002-999d00de-aa4343a5@192.168.0.93",LocalAddress="IPV4/UDP/192.168.0.189/5060",RemoteAddress="IPV4/UDP/192.168.0.93/49237",Challenge=""	
925	[2022-06-29 07:20:46] VERBOSE[2462] res_pjsip_logger.c: <--- Received SIP request (836 bytes) from UDP:192.168.0.93:49237 --->	
926	REGISTER sip:192.168.0.189 SIP/2.0	
927	Via: SIP/2.0/UDP 192.168.0.93:5060;branch=z9hG4bK1f7988f4	
928	From: <sip:302@192.168.0.189>;tag=e80462eb18ef0002b1f1d11c-ed92711b	
929	To: <sip:302@192.168.0.189>	
930	Call-ID: e80462eb-18ef0002-999d00de-aa4343a5@192.168.0.93	
931	Max-Forwards: 70	
932	Date: Wed, 29 Jun 2022 11:53:03 GMT	
933	CSeq: 102 REGISTER	
934	User-Agent: Cisco-CP7975G/8.5.3	
935	Contact: <sip:302@192.168.0.93:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-e80462eb18ef>";+u.sip!model.ccm.cisco.com="437"	
936	Authorization: Digest username="302",realm="asterisk",uri="sip:192.168.0.189",response="7c1b5c8f009a0acdeaf45cc723f62513",nonce="1656487246/5aaae3ec9b9267d8e034256bd9a3796a",opaque="12ed186c17af3e57",cnonce="c21c4d13",qop=auth,nc=00000001,algorithm=md5	
937	Supported: (null),X-cisco-xsi-7.0.1	
938	Content-Length: 0	
939	Expires: 3600	
940		
941		
942	[2022-06-29 07:20:46] VERBOSE[9425] res_pjsip_logger.c: <--- Transmitting SIP response (466 bytes) to UDP:192.168.0.93:5060 --->	
943	SIP/2.0 200 OK	
944	Via: SIP/2.0/UDP 192.168.0.93:5060;received=192.168.0.93;branch=z9hG4bK1f7988f4	
945	Call-ID: e80462eb-18ef0002-999d00de-aa4343a5@192.168.0.93	
946	From: <sip:302@192.168.0.189>;tag=e80462eb18ef0002b1f1d11c-ed92711b	
947	To: <sip:302@192.168.0.189>;tag=z9hG4bK1f7988f4	
948	CSeq: 102 REGISTER	
949	Date: Wed, 29 Jun 2022 07:20:46 GMT	
950	Contact: <sip:302@192.168.0.93:5060;transport=udp>;expires=3599	
951	Expires: 3600	
952	Server: FPBX-15.0.17.34(18.3.0)	
953	Content-Length: 0	
954		
955		
956	[2022-06-29 07:20:46] SECURITY[2514] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2022-06-29T07:20:46.664+0000",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="302",SessionID="e80462eb-18ef0002-999d00de-aa4343a5@192.168.0.93",LocalAddress="IPV4/UDP/192.168.0.189/5060",RemoteAddress="IPV4/UDP/192.168.0.93/49237",UsingPassword="1"	
957	[2022-06-29 07:20:46] VERBOSE[9425] res_pjsip_logger.c: <--- Transmitting SIP request (648 bytes) to UDP:192.168.0.93:5060 --->	
958	NOTIFY sip:302@192.168.0.93:5060;transport=udp SIP/2.0	
959	Via: SIP/2.0/UDP 192.168.0.189:5060;rport;branch=z9hG4bKPj683ff595-266c-4545-8989-ae4f2f219bc4	
960	From: <sip:302@192.168.0.189>;tag=d670e9ec-4b0c-41bf-80ea-f4e2f95c8256	
961	To: <sip:302@192.168.0.93>	
962	Contact: <sip:302@192.168.0.189:5060>	
963	Call-ID: 0e109830-8d53-4295-be1f-e4da438a810d	
964	CSeq: 41709 NOTIFY	
965	Subscription-State: terminated	
966	Event: message-summary	
967	Allow-Events: presence, dialog, message-summary, refer	
968	Max-Forwards: 70	
969	User-Agent: FPBX-15.0.17.34(18.3.0)	
970	Content-Type: application/simple-message-summary	
971	Content-Length: 48	
972		
973	Messages-Waiting: no	
974	Voice-Message: 0/0 (0/0)	
975		
976	[2022-06-29 07:20:46] VERBOSE[2462] res_pjsip_logger.c: <--- Received SIP response (337 bytes) from UDP:192.168.0.93:49293 --->	
977	SIP/2.0 200 OK	
978	Via: SIP/2.0/UDP 192.168.0.189:5060;rport;branch=z9hG4bKPj683ff595-266c-4545-8989-ae4f2f219bc4	
979	From: <sip:302@192.168.0.189>;tag=d670e9ec-4b0c-41bf-80ea-f4e2f95c8256	
980	To: <sip:302@192.168.0.93>	
981	Call-ID: 0e109830-8d53-4295-be1f-e4da438a810d	
982	Date: Wed, 29 Jun 2022 11:53:03 GMT	
983	CSeq: 41709 NOTIFY	
984	Content-Length: 0

I have NAT and Force port set to No.

Thanks

david551 · June 29, 2022, 1:23pm

It’s sending the wrong port number in the Via header. Given that, the Contact headers is also suspect.

There isn’t an option called NAT for chan_psjip. If you cannot fix the client or the router, you will need force rport and probably rewrite contact and symmetric media, turning on.

You are receiving from port 49237, but the Via header is claiming it was sent from 5060, and does not have rport set.

needmorebeerformewal · June 29, 2022, 2:51pm

Thanks David

Under advanced settings i have set the SIP Driver to be only chan_pjsip. I have also set under the extension rewrite contact to Yes and Force report to No.

It looks like its still trying to respond on the wrong port however and im not sure what i need to change to sort this.

2022-06-29 14:46:02] VERBOSE[12692] res_pjsip_logger.c: <--- Transmitting SIP request (424 bytes) to UDP:192.168.0.93:49157 --->	
43133	OPTIONS sip:302@192.168.0.93:49157 SIP/2.0	
43134	Via: SIP/2.0/UDP 192.168.0.189:5060;rport;branch=z9hG4bKPj4e71052f-82d1-42fc-a370-18dd4bff67e0	
43135	From: <sip:302@192.168.0.189>;tag=85682f6e-91f0-4464-96fa-868eac67ed23	
43136	To: <sip:302@192.168.0.93>	
43137	Contact: <sip:302@192.168.0.189:5060>	
43138	Call-ID: 5333b4f4-f40c-4210-8cdb-f2527d3d56ba	
43139	CSeq: 50414 OPTIONS	
43140	Max-Forwards: 70	
43141	User-Agent: FPBX-15.0.17.34(18.3.0)	
43142	Content-Length: 0

Its strange as it works fine on the software client just not the phone so im wondering if the issue is on the phone config side.

This is my SIPMac.cnf.xml file

<device>

<fullConfig>true</fullConfig>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>admin</sshUserId>

<sshPassword>cisco</sshPassword>

<sshAccess>0</sshAccess>

<sshPort>22</sshPort>

<devicePool>

<dateTimeSetting>

<dateTemplate>D.M.Y</dateTemplate>

<timeZone>GMT Standard/Daylight Time</timeZone>

<ntps>

<ntp>

<name>192.168.0.189</name>

<ntpMode>Unicast</ntpMode>

</ntp>

</ntps>

</dateTimeSetting>

<callManagerGroup>

<tftpDefault>true</tftpDefault>

<members>

<member priority="0">

<callManager>

<ports>

<ethernetPhonePort>2000</ethernetPhonePort>

<sipPort>5060</sipPort>

<securedSipPort>5061</securedSipPort>

</ports>

<processNodeName>192.168.0.189</processNodeName>

</callManager>

</member>

</members>

</callManagerGroup>

</devicePool>

<commonProfile>

<phonePassword>S@brina777</phonePassword>

<backgroundImageAccess>true</backgroundImageAccess>

<callLogBlfEnabled>2</callLogBlfEnabled>

</commonProfile>

<loadInformation>SIP75.8-5-4S</loadInformation>

<vendorConfig>

<disableSpeaker>false</disableSpeaker>

<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>

<pcPort>0</pcPort>

<settingsAccess>1</settingsAccess>

<garp>0</garp>

<voiceVlanAccess>0</voiceVlanAccess>

<videoCapability>0</videoCapability>

<autoSelectLineEnable>0</autoSelectLineEnable>

<daysDisplayNotActive>1</daysDisplayNotActive>

<displayOnTime>07:00</displayOnTime>

<displayOnDuration>10:02</displayOnDuration>

<displayIdleTimeout>00:05</displayIdleTimeout>

<webAccess>1</webAccess>

<spanToPCPort>1</spanToPCPort>

<loggingDisplay>1</loggingDisplay>

<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>

<loadServer></loadServer>

</vendorConfig>

<userLocale>

<name>English_United_Kingdom</name>

<uid>1</uid>

<langCode>en_GB</langCode>

<version>1.0.0.0-1</version>

<winCharSet>iso-8859-1</winCharSet>

</userLocale>

<networkLocale>United_Kingdom</networkLocale>

<networkLocaleInfo>

<name>United_Kingdom</name>

<uid>64</uid>

<version>1.0.0.0-1</version>

</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>

<authenticationURL>http://192.168.0.189/xmlservices//authentication.php</authenticationURL>

<directoryURL>http://192.168.0.189/xmlservices/E_book.php</directoryURL>

<idleTimeout>0</idleTimeout>

<idleURL></idleURL>

<informationURL></informationURL>

<proxyServerURL></proxyServerURL>

<servicesURL>http://cisco.internect.net/</servicesURL>

<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>

<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>

<dscpForCm2Dvce>96</dscpForCm2Dvce>

<transportLayerProtocol>2</transportLayerProtocol>

<capfAuthMode>0</capfAuthMode>

<capfList>

<capf>

<phonePort>3804</phonePort>

</capf>

</capfList>

<certHash></certHash>

<encrConfig>false</encrConfig>

<sipProfile>

<sipProxies>

<backupProxy>192.168.0.189</backupProxy>

<backupProxyPort>5060</backupProxyPort>

<emergencyProxy>192.168.0.189</emergencyProxy>

<emergencyProxyPort>5060</emergencyProxyPort>

<outboundProxy>192.168.0.189</outboundProxy>

<outboundProxyPort>5060</outboundProxyPort>

<registerWithProxy>true</registerWithProxy>

</sipProxies>

<sipCallFeatures>

<cnfJoinEnabled>true</cnfJoinEnabled>

<callForwardURI>x--serviceuri-cfwdall</callForwardURI>

<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>

<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>

<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>

<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>

<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>

<rfc2543Hold>true</rfc2543Hold>

<callHoldRingback>2</callHoldRingback>

<localCfwdEnable>true</localCfwdEnable>

<semiAttendedTransfer>false</semiAttendedTransfer>

<anonymousCallBlock>2</anonymousCallBlock>

<callerIdBlocking>0</callerIdBlocking>

<dndControl>0</dndControl>

<remoteCcEnable>true</remoteCcEnable>

</sipCallFeatures>

<sipStack>

<sipInviteRetx>6</sipInviteRetx>

<sipRetx>10</sipRetx>

<timerInviteExpires>180</timerInviteExpires>

<timerRegisterExpires>3600</timerRegisterExpires>

<timerRegisterDelta>5</timerRegisterDelta>

<timerKeepAliveExpires>120</timerKeepAliveExpires>

<timerSubscribeExpires>120</timerSubscribeExpires>

<timerSubscribeDelta>5</timerSubscribeDelta>

<timerT1>500</timerT1>

<timerT2>4000</timerT2>

<maxRedirects>70</maxRedirects>

<remotePartyID>false</remotePartyID>

<userInfo>None</userInfo>

</sipStack>

<autoAnswerTimer>1</autoAnswerTimer>

<autoAnswerAltBehavior>false</autoAnswerAltBehavior>

<autoAnswerOverride>true</autoAnswerOverride>

<transferOnhookEnabled>true</transferOnhookEnabled>

<enableVad>false</enableVad>

<preferredCodec>g711u</preferredCodec>

<dtmfAvtPayload>101</dtmfAvtPayload>

<dtmfDbLevel>3</dtmfDbLevel>

<dtmfOutofBand>avt</dtmfOutofBand>

<alwaysUsePrimeLine>true</alwaysUsePrimeLine>

<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>

<kpml>3</kpml>

<stutterMsgWaiting>1</stutterMsgWaiting>

<callStats>false</callStats>

<silentPeriodBetweenCallWaitingBursts>05</silentPeriodBetweenCallWaitingBursts>

<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

<startMediaPort>16384</startMediaPort>

<stopMediaPort>32766</stopMediaPort>

<voipControlPort>5060</voipControlPort>

<dscpForAudio>184</dscpForAudio>

<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>

<dialTemplate>dialplan.xml</dialTemplate>

<softKeyFile></softKeyFile>

<phoneLabel>302</phoneLabel>

<natEnabled></natEnabled>

<sipLines>

<line button="1">

<featureID>9</featureID>

<featureLabel>302</featureLabel>

<displayName>302</displayName>

<name>302</name>

<contact>302</contact>

<proxy>192.168.0.189</proxy>

<port>5060</port>

<autoAnswer>

<autoAnswerEnabled>2</autoAnswerEnabled>

</autoAnswer>

<callWaiting>3</callWaiting>

<authName>302</authName>

<authPassword>MyPassword</authPassword>

<sharedLine>false</sharedLine>

<messageWaitingLampPolicy>1</messageWaitingLampPolicy>

<messagesNumber>*97</messagesNumber>

<ringSettingIdle>4</ringSettingIdle>

<ringSettingActive>5</ringSettingActive>

<forwardCallInfoDisplay>

<callerName>true</callerName>

<callerNumber>false</callerNumber>

<redirectedNumber>true</redirectedNumber>

<dialedNumber>true</dialedNumber>

</forwardCallInfoDisplay>

</line>

</sipLines>

</sipProfile>

</device>

jcolp · June 29, 2022, 2:57pm

The “rewrite_contact” option doesn’t work with Cisco phones of that.

david551 · June 29, 2022, 2:58pm

force rport needs to be YES!

I believe that is currently the default. Although it creates a protocol violation, it would be unusual for there to be a situation where compliance was essential.

Unfortunately I’d forgotten that Cisco was an exception

jcolp · June 29, 2022, 2:59pm

Certain Cisco phones don’t actually work with rport. They won’t accept responses back on their source port, only in the port (5060) of the Via header. Same goes for Contact, has to be used as-is.

david551 · June 29, 2022, 3:01pm

Another possibility is that there is some sort of dynamic firewall rule being set up for the actual port used, but there is no way for the reply to get back to 5060.

david551 · June 29, 2022, 3:12pm

There is nothing in the log that indicates anything other than a successful registration.

The log is incomplete as there must have been a 401 response for it to get this far. That would imply that the 401 response got through, so why not the 200?

The Cisco is sending a garbled Supported header.

needmorebeerformewal · June 29, 2022, 3:49pm

I can see a 401 on line 58980.

I guess another question is does the Cisco 7975 actually work with Asterisk.

58962	[2022-06-29 15:46:59] SECURITY[18073] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2022-06-29T15:46:59.533+0000",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x30217e0",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/50332",UsingPassword="0",SessionTV="2022-06-29T15:46:59.533+0000"	
58963	[2022-06-29 15:46:59] VERBOSE[18025] res_pjsip_logger.c: <--- Received SIP request (582 bytes) from UDP:192.168.0.93:49219 --->	
58964	REGISTER sip:192.168.0.189 SIP/2.0	
58965	Via: SIP/2.0/UDP 192.168.0.93:5060;branch=z9hG4bKf346ada5	
58966	From: <sip:302@192.168.0.189>;tag=e80462eb18ef0012bb2bbfb7-388a32d1	
58967	To: <sip:302@192.168.0.189>	
58968	Call-ID: e80462eb-18ef0002-929ba012-39919b17@192.168.0.93	
58969	Max-Forwards: 70	
58970	Date: Wed, 29 Jun 2022 15:46:56 GMT	
58971	CSeq: 111 REGISTER	
58972	User-Agent: Cisco-CP7975G/8.5.3	
58973	Contact: <sip:302@192.168.0.93:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-e80462eb18ef>";+u.sip!model.ccm.cisco.com="437"	
58974	Supported: (null),X-cisco-xsi-7.0.1	
58975	Content-Length: 0	
58976	Expires: 3600	
58977		
58978		
58979	[2022-06-29 15:46:59] VERBOSE[18026] res_pjsip_logger.c: <--- Transmitting SIP response (517 bytes) to UDP:192.168.0.93:49219 --->	
58980	SIP/2.0 401 Unauthorized	
58981	Via: SIP/2.0/UDP 192.168.0.93:5060;rport=49219;received=192.168.0.93;branch=z9hG4bKf346ada5	
58982	Call-ID: e80462eb-18ef0002-929ba012-39919b17@192.168.0.93	
58983	From: <sip:302@192.168.0.189>;tag=e80462eb18ef0012bb2bbfb7-388a32d1	
58984	To: <sip:302@192.168.0.189>;tag=z9hG4bKf346ada5	
58985	CSeq: 111 REGISTER	
58986	WWW-Authenticate: Digest realm="asterisk",nonce="1656517619/00885a667b48d88e1aca2512ee2aa62a",opaque="0f1be94a76636353",algorithm=md5,qop="auth"	
58987	Server: FPBX-15.0.17.34(18.3.0)	
58988	Content-Length: 0	
58989		
58990		
58991	[2022-06-29 15:46:59] SECURITY[18073] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2022-06-29T15:46:59.641+0000",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="302",SessionID="e80462eb-18ef0002-929ba012-39919b17@192.168.0.93",LocalAddress="IPV4/UDP/192.168.0.189/5060",RemoteAddress="IPV4/UDP/192.168.0.93/49219",Challenge=""	
58992	[2022-06-29 15:47:02] SECURITY[18073] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2022-06-29T15:47:02.756+0000",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x7f12300029f0",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/50338",UsingPassword="0",SessionTV="2022-06-29T15:47:02.756+0000"

needmorebeerformewal · June 29, 2022, 3:50pm

Do you know if the 7975 is one of them?

jcolp · June 29, 2022, 3:51pm

I don’t for sure.

needmorebeerformewal · June 30, 2022, 6:20am

From what i have read it does work.

Will keep playing and see if i can resolve the issue.

needmorebeerformewal · June 30, 2022, 7:56am

Just been playing around with it some more and if i use Zoiper using the same account details it logs in fine. I can see on line 222 it is also adding a random port.

222	[2022-06-29 18:59:36] VERBOSE[5629] res_pjsip_logger.c: <--- Received SIP request (863 bytes) from UDP:192.168.0.76:56920 --->	
223	REGISTER sip:192.168.0.189;transport=UDP SIP/2.0	
224	Via: SIP/2.0/UDP 192.168.0.76:56920;branch=z9hG4bK-524287-1---e07f2ed101e5c47e;rport	
225	Max-Forwards: 70	
226	Contact: <sip:302@192.168.0.76:56920;transport=UDP;rinstance=fa344a49cb71251f>;expires=0	
227	To: <sip:302@192.168.0.189;transport=UDP>	
228	From: <sip:302@192.168.0.189;transport=UDP>;tag=d6179207	
229	Call-ID: CuaZ4hZHNYjbv9JRpHmGuQ..	
230	CSeq: 3 REGISTER	
231	Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE	
232	User-Agent: Z 5.5.12 v2.10.18.2	
233	Authorization: Digest username="302",realm="asterisk",nonce="1656529173/fd4a9d9b5d5202edccb6fb1336f43700",uri="sip:192.168.0.189;transport=UDP",response="12dd6f6385cb2bf589b44aa12de2d08b",cnonce="3d03d22c437fdfdfe673ee41c99b602e",nc=00000002,qop=auth,algorithm=md5,opaque="7744bfd13d416bd7"	
234	Allow-Events: presence, kpml, talk	
235	Content-Length: 0

If i trying from the 7975 i get the following.

I notice on lines 2747 and 2748 the tag is not UDP but a string so im not sure where it is getting that from or if that is the problem.


[2022-06-29 19:03:48] VERBOSE[20393] res_pjsip_logger.c: <--- Transmitting SIP response (517 bytes) to UDP:192.168.0.93:49156 --->	
2744	SIP/2.0 401 Unauthorized	
2745	Via: SIP/2.0/UDP 192.168.0.93:5060;rport=49156;received=192.168.0.93;branch=z9hG4bK0d65caa2	
2746	Call-ID: e80462eb-18ef0002-4c63c5e0-3dcbf19e@192.168.0.93	
2747	From: <sip:302@192.168.0.189>;tag=e80462eb18ef000e9882b1fb-244bf828	
2748	To: <sip:302@192.168.0.189>;tag=z9hG4bK0d65caa2	
2749	CSeq: 113 REGISTER	
2750	WWW-Authenticate: Digest realm="asterisk",nonce="1656529428/1690377af2970a91c92d1a9bb5210cda",opaque="0ad7a6726e80c874",algorithm=md5,qop="auth"	
2751	Server: FPBX-15.0.17.34(18.3.0)	
2752	Content-Length: 0	
2753		
2754		
2755	[2022-06-29 19:03:48] SECURITY[5677] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2022-06-29T19:03:48.523+0000",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="302",SessionID="e80462eb-18ef0002-4c63c5e0-3dcbf19e@192.168.0.93",LocalAddress="IPV4/UDP/192.168.0.189/5060",RemoteAddress="IPV4/UDP/192.168.0.93/49156",Challenge=""	
2756	[2022-06-29 19:03:49] VERBOSE[5629] res_pjsip_logger.c: <--- Received SIP request (685 bytes) from UDP:192.168.0.93:49156 --->	
2757	REGISTER sip:192.168.0.189 SIP/2.0	
2758	Via: SIP/2.0/UDP 192.168.0.93:5060;branch=z9hG4bK0d65caa2	
2759	From: <sip:302@192.168.0.189>;tag=e80462eb18ef000e9882b1fb-244bf828	
2760	To: <sip:302@192.168.0.189>	
2761	Call-ID: e80462eb-18ef0002-4c63c5e0-3dcbf19e@192.168.0.93	
2762	Max-Forwards: 70	
2763	Date: Wed, 16 Dec 2009 08:52:39 GMT	
2764	CSeq: 113 REGISTER	
2765	User-Agent: Cisco-CP7975G/8.5.3	
2766	Contact: <sip:302@192.168.0.93:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-e80462eb18ef>";+u.sip!model.ccm.cisco.com="437"	
2767	Supported: (null),X-cisco-xsi-7.0.1	
2768	Content-Length: 0	
2769	Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEPE80462EB18EF Load=term75.default Last=initialized"	
2770	Expires: 3600

david551 · June 30, 2022, 8:39am

tag is being used correctly, and should be a random string.

The difference is that the Via in Zoiper trace matches the source IP address.

However, Joshua has pointed out that the mismatch in Cisco’s is deliberate,and they expect the Via to be obeyed. The confusing bit is that the Cisco seems to be happy with the 401, but, according to you, is ignoring the 200, even though they are routed the same. As far as Asterisk is concerned, the Cisco is registered, and that should be all that is actually needed for calls to work both ways.

As Asterisk is accepting the registration, the lack of an explicit transport option clearly isn’t a problem.

The “(null)” in the Supported header clearly shows that the Cisco firmware is buggy.

needmorebeerformewal · June 30, 2022, 8:46am

I will try a different firmware and see if there are any changes in response.

I was not sure if the To and From should contain UDP like Zoiper or not on the Cisco connection attempt.

needmorebeerformewal · July 1, 2022, 9:50am

In order get the cisco 7975 to work with Asterisk you need to install chan_sccp and create users under the sip driver type. You will also need to install the sccp_manager.

After a lot of playing whilst you can get it to work under pjsip some of the functions to do not work and it is prone to erroring – SCCP is the most stable route I have found so far.

I have put all the files needed on my github so hopefully this can help someone else facing the same problem.

Install guide

Extract the Cisco7975TFTPConfig.zip Update the SEPMACADDRESS.cnf.xml to be SEPyourmacaddress.cnf.xml of using the mac address of the phone.

Update SEPyourmacaddress.cnf.xml with the details of your sip server

Upload files to your TFTP server and update the Cisco phone

Please note : if in settings of the phone type # to reset To factory reset power on the phone and when you see the first light appear hold # and then when the lights start to flow type 123456789*0#

SSH onto your voip server and run the following.

Yum install git mc nano asterisk-devel

cd /usr/src/ git clone GitHub - chan-sccp/chan-sccp: Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. cd chan-sccp

./configure --enable-conference make -j2 make install

vim /etc/asterisk/modules.conf noload => chan_skinny.so load => chan_sccp.so

cp sccp.conf.freepbx /etc/asterisk/sccp.conf cp sccp_extensions.conf.freepbx /etc/asterisk/sccp_extensions.conf cp sccp_hardware.conf.freepbx /etc/asterisk/sccp_hardware.conf

fwconsole restart

Once restarted in Asterisk under Admin – Module Admin upload and install the sccp_manager-14-2-0.11.zip

nrjchnd · July 1, 2022, 12:15pm

You can also use a patched Chan_sip module to run the sip firmware on the cisco phones (including video variants)

Have you considered:

https://usecallmanager.nz/documentation-overview.html
by Gareth Palmer.

Have been using this for years now.

system · July 31, 2022, 12:15pm

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