Cisco IP Phone I can't call!

Hello everyone!
I have Asterisk server and cisco ip phone 7965 and all file of sip protocol and cnf.xml and xml was created with information, In this time i can make call from zoiper or 3cx software to cisco phone without any problem and i can listen between them, But the problem when I need call from cisco phone to zoiper or 3cx I cant because direct the line go to like disconnect the line by beep!!! What the problem or how can I solve it please !??
debug
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f50600235d0 – Strict RTP learning after remote address set to: zoiper ip

Thank you

Are you using the sip or pjsip channel technologies?

You need to provide the protocol logging for whichever of the above you are using.

You may also need to provide the relevant configuration file, if you want more than general hints at what is wrong.

Note Asterisk doesn’t send beeps as the result of a failed call, unless there is explicit dialplan to do so, or the called party sends one in the media stream. It will simply send a SIP failure response, and the phone may or may not generate some form of sound as a result.

I am used SIP
cnf.xml file

<?xml version="1.0" encoding="UTF-8"?>
<device>
    <fullConfig>true</fullConfig>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>admin</sshUserId>
    <sshPassword>pass</sshPassword>
    <devicePool>
        <dateTimeSetting>
            <dateTemplate>D-M-Ya</dateTemplate>
 
    <timeZone>Eastern Standard/Daylight Time</timeZone>
                    <ntps>
                <ntp>
                    <name>91.213.191.21</name>
      				<ntpMode>unicast</ntpMode>
    			</ntp>
            </ntps>
                    </dateTimeSetting>
        <callManagerGroup>
            <members>
                <member priority="0">
                    <callManager>
                        <name>1.1.1.200</name>
						<description></description>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                            <sipPort>5160</sipPort>
                            <securedSipPort>5061</securedSipPort>
                        </ports>
                        <processNodeName>1.1.1.200</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>
				<connectionMonitorDuration>120</connectionMonitorDuration>
	</devicePool>
    <commonProfile>
        <phonePassword></phonePassword>
        <backgroundImageAccess>true</backgroundImageAccess>
        <callLogBlfEnabled>1</callLogBlfEnabled>
    </commonProfile>
    <loadInformation></loadInformation>
    <vendorConfig>
        <disableSpeaker>false</disableSpeaker>
        <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
        <pcPort>0</pcPort>
        <settingsAccess>1</settingsAccess>
        <garp>0</garp>
        <voiceVlanAccess>0</voiceVlanAccess>
        <g722CodecSupport>2</g722CodecSupport>
        <handsetWidebandEnable>1</handsetWidebandEnable>
        <headsetWidebandEnable>0</headsetWidebandEnable>
        <headsetWidebandUIControl>0</headsetWidebandUIControl>
        <handsetWidebandUIControl>0</handsetWidebandUIControl>
        <videoCapability>0</videoCapability>
        <autoSelectLineEnable>0</autoSelectLineEnable>
        <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
        <displayOnTime></displayOnTime>
        <displayOnDuration></displayOnDuration>
        <displayIdleTimeout>00:05</displayIdleTimeout>
        <webAccess>0</webAccess>
        <spanToPCPort>0</spanToPCPort>
        <loggingDisplay>1</loggingDisplay>
        <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
        <loadServer></loadServer>
    </vendorConfig>
   <userLocale>
 <name>English_United_States</name>
<uid>1</uid>
 <langCode>en_US</langCode>
<version>1.0.0.0-1</version>
 <winCharSet>iso-8859-1</winCharSet>
</userLocale>
    <networkLocale>United_States</networkLocale>
<networkLocaleInfo>
 <name>United_States</name>

 <version>1.0.0.0-1</version>
</networkLocaleInfo>
    <deviceSecurityMode>1</deviceSecurityMode>
    <authenticationURL></authenticationURL>
    <directoryURL></directoryURL>
    <idleTimeout>0</idleTimeout>
    <idleURL></idleURL>
    <informationURL></informationURL>
    <messagesURL></messagesURL>
    <proxyServerURL></proxyServerURL>
    <servicesURL></servicesURL>
    <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
    <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
    <dscpForCm2Dvce>96</dscpForCm2Dvce>
    <transportLayerProtocol>2</transportLayerProtocol>
    <dndCallAlert>5</dndCallAlert>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
        <capf>
            <phonePort>3804</phonePort>
        </capf>
    </capfList>
    <certHash></certHash>
    <encrConfig>false</encrConfig>
    <sipProfile>
        <sipProxies>
            <backupProxy></backupProxy>
            <backupProxyPort>5060</backupProxyPort>
            <emergencyProxy></emergencyProxy>
            <emergencyProxyPort>5060</emergencyProxyPort>
            <outboundProxy></outboundProxy>
            <outboundProxyPort>5060</outboundProxyPort>
            <registerWithProxy>true</registerWithProxy>
        </sipProxies>
        <sipCallFeatures>
            <cnfJoinEnabled>true</cnfJoinEnabled>
			<callForwardURI>x-cisco-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>false</localCfwdEnable>
            <semiAttendedTransfer>true</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>1200</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>g711alaw</preferredCodec>
        <dtmfAvtPayload>101</dtmfAvtPayload>
        <dtmfDbLevel>3</dtmfDbLevel>
        <dtmfOutofBand>avt</dtmfOutofBand>
        <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
        <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
        <kpml>3</kpml>
        <stutterMsgWaiting>0</stutterMsgWaiting>
        <callStats>false</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
        <startMediaPort>16384</startMediaPort>
        <stopMediaPort>32766</stopMediaPort>
        <voipControlPort>5060</voipControlPort>
        <dscpForAudio>184</dscpForAudio>
        <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
        <dialTemplate>cisco_dialplan.xml</dialTemplate>
        <softKeyFile>softkeys.xml</softKeyFile>
        <phoneLabel>400</phoneLabel>
        <natEnabled>false</natEnabled>

        <sipLines>
            <line button="1">
                <featureID>9</featureID>
                <featureLabel>Cisco IP phone 7965</featureLabel>
                <name>400</name>
                <displayName>400</displayName>
                <contact>400</contact>
                <proxy>USECALLMANAGER</proxy>
                <port>5160</port>
                <autoAnswer>
                    <autoAnswerEnabled>2</autoAnswerEnabled>
                </autoAnswer>
                <callWaiting>1</callWaiting>
                <authName>400</authName>
                <authPassword>12345678</authPassword>
                <sharedLine>false</sharedLine>
                <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
                <messagesNumber>*97</messagesNumber>
                <ringSettingIdle>4</ringSettingIdle>
                <ringSettingActive>5</ringSettingActive>
                <forwardCallInfoDisplay>
                    <callerName>true</callerName>
                    <callerNumber>false</callerNumber>
                    <redirectedNumber>false</redirectedNumber>
                    <dialedNumber>true</dialedNumber>
                </forwardCallInfoDisplay>
            </line>
		
            </sipLines>
    </sipProfile>
</device>

And other xml file

<DIALTEMPLATE>

<TEMPLATE User="Phone" TIMEOUT="2" MATCH="*"/>

<TEMPLATE TIMEOUT="0" MATCH="*#" REWRITE="%1"/>

</DIALTEMPLATE>

I am used SIP
Cisco ip phone account:
Ext. 400
pass. 12345678
Asterisk server: 1.1.1.200

<?xml version="1.0" encoding="UTF-8"?> true SIP admin pass D-M-Ya
<timeZone>Eastern Standard/Daylight Time</timeZone>
                <ntps>
            <ntp>
                <name>91.213.191.21</name>
  				<ntpMode>unicast</ntpMode>
			</ntp>
        </ntps>
                </dateTimeSetting>
    <callManagerGroup>
        <members>
            <member priority="0">
                <callManager>
                    <name>1.1.1.200</name>
					<description></description>
                    <ports>
                        <ethernetPhonePort>2000</ethernetPhonePort>
                        <sipPort>5160</sipPort>
                        <securedSipPort>5061</securedSipPort>
                    </ports>
                    <processNodeName>1.1.1.200</processNodeName>
                </callManager>
            </member>
        </members>
    </callManagerGroup>
			<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<commonProfile>
    <phonePassword></phonePassword>
    <backgroundImageAccess>true</backgroundImageAccess>
    <callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<loadInformation></loadInformation>
<vendorConfig>
    <disableSpeaker>false</disableSpeaker>
    <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
    <pcPort>0</pcPort>
    <settingsAccess>1</settingsAccess>
    <garp>0</garp>
    <voiceVlanAccess>0</voiceVlanAccess>
    <g722CodecSupport>2</g722CodecSupport>
    <handsetWidebandEnable>1</handsetWidebandEnable>
    <headsetWidebandEnable>0</headsetWidebandEnable>
    <headsetWidebandUIControl>0</headsetWidebandUIControl>
    <handsetWidebandUIControl>0</handsetWidebandUIControl>
    <videoCapability>0</videoCapability>
    <autoSelectLineEnable>0</autoSelectLineEnable>
    <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
    <displayOnTime></displayOnTime>
    <displayOnDuration></displayOnDuration>
    <displayIdleTimeout>00:05</displayIdleTimeout>
    <webAccess>0</webAccess>
    <spanToPCPort>0</spanToPCPort>
    <loggingDisplay>1</loggingDisplay>
    <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
    <loadServer></loadServer>
</vendorConfig>
English_United_States 1 en_US 1.0.0.0-1 iso-8859-1 United_States United_States

1.0.0.0-1

1


0





96
0
96
2
5
0


3804



false



5060

5060

5060
true


true
x-cisco-serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
true
2
false
true
2
0
0
true


6
10
180
1200
5
120
120
5
500
4000
70
false
None

1
false
true
true
false
g711alaw
101
3
avt
false
false
3
0
false
10
false
16384
32766
5060
184
0
cisco_dialplan.xml
softkeys.xml
400
false

    <sipLines>
        <line button="1">
            <featureID>9</featureID>
            <featureLabel>Cisco IP phone 7965</featureLabel>
            <name>400</name>
            <displayName>400</displayName>
            <contact>400</contact>
            <proxy>USECALLMANAGER</proxy>
            <port>5160</port>
            <autoAnswer>
                <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>1</callWaiting>
            <authName>400</authName>
            <authPassword>12345678</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <forwardCallInfoDisplay>
                <callerName>true</callerName>
                <callerNumber>false</callerNumber>
                <redirectedNumber>false</redirectedNumber>
                <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
        </line>
	
        </sipLines>
</sipProfile>

other file xml

		<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<commonProfile>
    <phonePassword></phonePassword>
    <backgroundImageAccess>true</backgroundImageAccess>
    <callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<loadInformation></loadInformation>
<vendorConfig>
    <disableSpeaker>false</disableSpeaker>
    <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
    <pcPort>0</pcPort>
    <settingsAccess>1</settingsAccess>
    <garp>0</garp>
    <voiceVlanAccess>0</voiceVlanAccess>
    <g722CodecSupport>2</g722CodecSupport>
    <handsetWidebandEnable>1</handsetWidebandEnable>
    <headsetWidebandEnable>0</headsetWidebandEnable>
    <headsetWidebandUIControl>0</headsetWidebandUIControl>
    <handsetWidebandUIControl>0</handsetWidebandUIControl>
    <videoCapability>0</videoCapability>
    <autoSelectLineEnable>0</autoSelectLineEnable>
    <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
    <displayOnTime></displayOnTime>
    <displayOnDuration></displayOnDuration>
    <displayIdleTimeout>00:05</displayIdleTimeout>
    <webAccess>0</webAccess>
    <spanToPCPort>0</spanToPCPort>
    <loggingDisplay>1</loggingDisplay>
    <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
    <loadServer></loadServer>
</vendorConfig>
English_United_States 1 en_US 1.0.0.0-1 iso-8859-1 United_States United_States

1.0.0.0-1

1


0





96
0
96
2
5
0


3804



false



5060

5060

5060
true


true
x-cisco-serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
true
2
false
true
2
0
0
true


6
10
180
1200
5
120
120
5
500
4000
70
false
None

1
false
true
true
false
g711alaw
101
3
avt
false
false
3
0
false
10
false
16384
32766
5060
184
0
cisco_dialplan.xml
softkeys.xml
400
false

    <sipLines>
        <line button="1">
            <featureID>9</featureID>
            <featureLabel>Cisco IP phone 7965</featureLabel>
            <name>400</name>
            <displayName>400</displayName>
            <contact>400</contact>
            <proxy>USECALLMANAGER</proxy>
            <port>5160</port>
            <autoAnswer>
                <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>1</callWaiting>
            <authName>400</authName>
            <authPassword>12345678</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <forwardCallInfoDisplay>
                <callerName>true</callerName>
                <callerNumber>false</callerNumber>
                <redirectedNumber>false</redirectedNumber>
                <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
        </line>
	
        </sipLines>
</sipProfile>

As you say you are using chan_sip, we need the output from “sip set debug on” and the contents of sip.conf and anything that it includes, in as much as it is in the general section or relates to any of the parties in the call.

I am used command “asterisk -r” then used “sip set debug on” and the result show below:


<--- SIP read from UDP:1.1.1.56:53100 --->


<------------->

<--- SIP read from UDP:1.1.1.50:51484 --->
INVITE sip:2@1.1.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.50:5060;branch=z9hG4bK999aa8c7
From: "400 - Ayad Walid" <sip:400@1.1.1.200>;tag=d4d74841b28c0008c4d84b9e-e5d370                                                                                                                81
To: <sip:2@1.1.1.200;user=phone>
Call-ID: d4d74841-b28c0007-aa872600-020aff1b@1.1.1.50
Max-Forwards: 70
Date: Fri, 18 Sep 2020 06:18:14 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7965G/9.2.1
Contact: <sip:400@1.1.1.50:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c                                                                                                                isco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallbac                                                                                                                k,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 344
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 21990 0 IN IP4 1.1.1.50
s=SIP Call
t=0 0
m=audio 23910 RTP/AVP 8 0 18 102 116 101
c=IN IP4 1.1.1.50
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 16 lines) ---
Sending to 1.1.1.50:5060 (no NAT)
Sending to 1.1.1.50:5060 (no NAT)
Using INVITE request as basis request - d4d74841-b28c0007-aa872600-020aff1b@1.1.                                                                                                                1.50
Found peer '400' for '400' from 1.1.1.50:51484

<--- Reliably Transmitting (no NAT) to 1.1.1.50:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 1.1.1.50:5060;branch=z9hG4bK999aa8c7;received=1.1.1.50
From: "400 - Ayad Walid" <sip:400@1.1.1.200>;tag=d4d74841b28c0008c4d84b9e-e5d370                                                                                                                81
To: <sip:2@1.1.1.200;user=phone>;tag=as268ba1f7
Call-ID: d4d74841-b28c0007-aa872600-020aff1b@1.1.1.50
CSeq: 101 INVITE
Server: FPBX-15.0.16.42(16.6.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                                                H, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06b8f659"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'd4d74841-b28c0007-aa872600-020aff1b@1.1.1.                                                                                                                50' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:1.1.1.50:50542 --->
ACK sip:2@1.1.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.50:5060;branch=z9hG4bK999aa8c7
From: "400 - Ayad Walid" <sip:400@1.1.1.200>;tag=d4d74841b28c0008c4d84b9e-e5d370                                                                                                                81
To: <sip:2@1.1.1.200;user=phone>;tag=as268ba1f7
Call-ID: d4d74841-b28c0007-aa872600-020aff1b@1.1.1.50
Date: Fri, 18 Sep 2020 06:18:14 GMT
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:1.1.1.50:51484 --->
INVITE sip:2@1.1.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.50:5060;branch=z9hG4bK5ffc804c
From: "400 - Ayad Walid" <sip:400@1.1.1.200>;tag=d4d74841b28c0008c4d84b9e-e5d370                                                                                                                81
To: <sip:2@1.1.1.200;user=phone>
Call-ID: d4d74841-b28c0007-aa872600-020aff1b@1.1.1.50
Max-Forwards: 70
Date: Fri, 18 Sep 2020 06:18:14 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7965G/9.2.1
Contact: <sip:400@1.1.1.50:5060;transport=udp>
Authorization: Digest username="400",realm="asterisk",uri="sip:2@1.1.1.200;user=                                                                                                                phone",response="f3f77de0b8da7274b5cb011cc5148b68",nonce="06b8f659",algorithm=MD                                                                                                                5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-c                                                                                                                isco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallbac                                                                                                                k,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 344
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 21990 0 IN IP4 1.1.1.50
s=SIP Call
t=0 0
m=audio 23910 RTP/AVP 8 0 18 102 116 101
c=IN IP4 1.1.1.50
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 16 lines) ---
Sending to 1.1.1.50:5060 (no NAT)
Using INVITE request as basis request - d4d74841-b28c0007-aa872600-020aff1b@1.1.                                                                                                                1.50
Found peer '400' for '400' from 1.1.1.50:51484
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw|g729|slin16|ilbc)/video=                                                                                                                (nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon                                                                                                                e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.50:23910
Looking for 2 in from-internal (domain 1.1.1.200)

<--- Reliably Transmitting (no NAT) to 1.1.1.50:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 1.1.1.50:5060;branch=z9hG4bK5ffc804c;received=1.1.1.50
From: "400 - Ayad Walid" <sip:400@1.1.1.200>;tag=d4d74841b28c0008c4d84b9e-e5d370                                                                                                                81
To: <sip:2@1.1.1.200;user=phone>;tag=as268ba1f7
Call-ID: d4d74841-b28c0007-aa872600-020aff1b@1.1.1.50
CSeq: 102 INVITE
Server: FPBX-15.0.16.42(16.6.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                                                H, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'd4d74841-b28c0007-aa872600-020aff1b@1.1.1.                                                                                                                50' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:1.1.1.50:50943 --->
ACK sip:2@1.1.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.50:5060;branch=z9hG4bK5ffc804c
From: "400 - Ayad Walid" <sip:400@1.1.1.200>;tag=d4d74841b28c0008c4d84b9e-e5d370                                                                                                                81
To: <sip:2@1.1.1.200;user=phone>;tag=as268ba1f7
Call-ID: d4d74841-b28c0007-aa872600-020aff1b@1.1.1.50
Date: Fri, 18 Sep 2020 06:18:14 GMT
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'UkrFo5kQBIuQLZqh1xov2g..' Method: REGISTER
Really destroying SIP dialog 'ZWI3NjVlMzk4YTdlNWQ0YWRmYmQyZDRjOGM5YjQ0Y2Q.' Method: REGISTER
freepbx*CLI>

Please advise me for how to solve the problem!!!

from your logs, its clear that you are trying to dial non-existing extension “2” from 400 ,

In this time I solved some problem, Now i can make call from cisco to zoiper or other one like 3cx, So If i call from cisco to zoiper I can listen inside of zoiper perfect but cisco phone I cant listen !!! why ? I dont now!!
So in the same idea but if I call from cisco phone to 3cx the line between them its very good without any problem because in to sides can listen and Speak up between them, I tried to check the codec in server and zoiper and I enable all of zoiper codec inside of server but the problem not solve??
Any help about the new problem?
Thank you

enable option nat on your freePBX for both extensions.

net=yes
or
nat=force_rport,comedia

I am tried now so the result:
In cisco phone when I dial to zoiper the cisco open the line but the connection is still open in cisco and the other party is not contacted or ring after 1 mint. the line go to disconnected !!!

Again you need to provide logs.

Also note that nat= does not enable NAT handling in Asterisk; what it does is to enable workarounds for some cases where NAT is not properly handled, in particular, where it is not properly handled by the remote end. It doesn’t make Asterisk handle NAT properly when Asterisk is inside NAT.

The IP addresses in the logs are either obfuscated or being misused, so the logs don’t tell us whether NAT is actually being used.

I know should the NAT be = no “off”, So what you need about log please write me the “command cli” ?

nat= should be left at default in most cases. The default is not off.

Setting it to off disables the automatic use of NAT handling features when the real NAT settings (e.g. localnet and externip) indicate NAT is being used.

You need to describe the structure of your network, and particular the location of any address translation.

You should also note that you should not, normally, be using chan_sip for new systems, as it is only community supported, so bugs will take a long time to fix, if at all.

Yes, Thanks for your information, So what you need me for logs please write me the command after that I can get from my server!

The same as before, but on a call that is failing in the new way. Also you need to explain the 1.1.1. in your addresses. It is particularly important to distinguish between public and private addresses.

For the network structure, there is no command; you need to understand how your network is put together.

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