404 error when trying to call extension from another extension

Hi All, I have been struggling with this for a few weeks now, I have read all I can find on the 404 error but everything I can find seems related to incoming from providers not other extensions

Note (all ip addresses have been changed for security)
Below is the cli debug from the extension in question, it is extension 103, the call was placed from ex 101.

Calls from ex 101 to other extensions or external work fine
Nothing can place a call to 103 internal or external

The setup is freepbx is on a static public IP
I have opened and forwarded all recomended ports I can find.

Both extensions are behind dls nat routers, ex 101 is also a 7940 that is working fine so I believe the basic config is working ok

The 404 seems to be from the phone (cisco 7940), only other thing I can see in the log is a congestion problem?? But not sure how to interpret them really

This is the line info from the 7940 at ex103

SIP Configuration Generic File

Line 1 appearance

line1_name: "103"
line1_shortname : “Resource 103”

Line 1 Registration Authentication

line1_authname: “103”

Line 1 Registration Password

line1_password: "xxxxxx"
line1_contact : “103”

the phone appears to be registered ok
Name/username Host Dyn Nat ACL Port Status
103/103 213.121.131.184 D N A 5070 OK (309 ms)
110/110 81.131.84.235 D N A 5060 OK (19 ms)

101/101 81.153.185.96 D N A 5070 OK (319 ms)

Any pointers or guidance or help would be greatly appreciated.
let me know any othe info that may help
I believe the call is making it to the phone but the phone is rejecting it ??

Thanks Paul

-- Executing [[email protected]:1] e[1;36;40mMacroe[0;37;40m("e[1;35;40mSIP/101-097fea60e[0;37;40m", "e[1;35;40mexten-vm|novm|103e[0;37;40m") in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:1] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40muser-calleride[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:1] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mAMPUSER=101e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:2] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?reporte[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:3] e[1;36;40mExecIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1|Set|REALCALLERIDNUM=101e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:4] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mAMPUSER=101e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:5] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mAMPUSERCIDNAME=Julia Resourcee[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:6] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?reporte[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:7] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mAMPUSERCID=101e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:8] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mCALLERID(all)=“Julia Resource” <101>e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:9] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mREALCALLERIDNUM=101e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:10] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?continuee[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:11] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m__TTL=64e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:12] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1?continuee[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Goto (macro-user-callerid,s,19)

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:19] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mUsing CallerID “Julia Resource” <101>e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:2] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mRingGroupMethod=nonee[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:3] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mVMBOX=novme[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:4] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mEXTTOCALL=103e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:5] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mCFUEXT=e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:6] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mCFBEXT=e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:7] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mRT=”“e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:8] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mrecord-enable|103|INe[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:1] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1?checke[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Goto (macro-record-enable,s,4)

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:4] e[1;36;40mAGIe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mrecordingcheck|20090516-161512|1242486912.72e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

e[Khost81-138-84-236*CLI>
recordingcheck|20090516-161512|1242486912.72: Inbound recording not enabled

e[Khost81-138-84-236*CLI>
– AGI Script recordingcheck completed, returning 0

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:5] e[1;36;40mMacroExite[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:9] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mdial||tr|103e[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:1] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1?diale[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Goto (macro-dial,s,3)

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:3] e[1;36;40mAGIe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mdialparties.agie[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi

e[Khost81-138-84-236*CLI>
dialparties.agi: Starting New Dialparties.agi

e[Khost81-138-84-236*CLI>
== Parsing ‘/etc/asterisk/manager.conf’: Found

e[Khost81-138-84-236*CLI>
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found

e[Khost81-138-84-236*CLI>
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found

e[Khost81-138-84-236*CLI>
== Manager ‘admin’ logged on from 127.0.0.1

e[Khost81-138-84-236*CLI>
dialparties.agi: Caller ID name is ‘Julia Resource’ number is ‘101’

e[Khost81-138-84-236*CLI>
dialparties.agi: USE_CONFIRMATION: ‘FALSE’

e[Khost81-138-84-236*CLI>
dialparties.agi: RINGGROUP_INDEX: ‘’

e[Khost81-138-84-236*CLI>
dialparties.agi: Methodology of ring is ‘none’

e[Khost81-138-84-236*CLI>
– dialparties.agi: Added extension 103 to extension map

e[Khost81-138-84-236*CLI>
– dialparties.agi: Extension 103 cf is disabled

e[Khost81-138-84-236*CLI>
– dialparties.agi: Extension 103 do not disturb is disabled

e[Khost81-138-84-236*CLI>
> dialparties.agi: extnum 103 has: cw: 1; hascfb: 0 [] hascfu: 0 []

e[Khost81-138-84-236*CLI>
> dialparties.agi: ExtensionState: 0

e[Khost81-138-84-236*CLI>
– dialparties.agi: dbset CALLTRACE/103 to 101

e[Khost81-138-84-236*CLI>
– dialparties.agi: Filtered ARG3: 103

e[Khost81-138-84-236*CLI>
== Manager ‘admin’ logged off from 127.0.0.1

e[Khost81-138-84-236*CLI>
– AGI Script dialparties.agi completed, returning 0

e[Khost81-138-84-236*CLI>
– Executing [[email protected]:7] e[1;36;40mDiale[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mSIP/103||tre[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
Audio is at 81.138.84.236 port 18452

e[Khost81-138-84-236*CLI>
Adding codec 0x4 (ulaw) to SDP

e[Khost81-138-84-236*CLI>
Adding codec 0x8 (alaw) to SDP

e[Khost81-138-84-236*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

e[Khost81-138-84-236*CLI>
Reliably Transmitting (NAT) to 213.121.131.184:5070:
INVITE sip:[email protected]:5070;transport=udp SIP/2.0

Via: SIP/2.0/UDP 81.138.84.236:5060;branch=z9hG4bK1bdff7dd;rport

From: “Julia Resource” sip:[email protected];tag=as7529094e

To: sip:[email protected]:5070;transport=udp

Contact: sip:[email protected]

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Sat, 16 May 2009 15:15:12 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 264

v=0

o=root 2776 2776 IN IP4 81.138.84.236

s=session

c=IN IP4 81.138.84.236

t=0 0

m=audio 18452 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


e[Khost81-138-84-236*CLI>
– Called 103

e[Khost81-138-84-236*CLI>

<— SIP read from 213.121.131.184:5070 —>
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 81.138.84.236:5060;branch=z9hG4bK1bdff7dd;rport

From: “Julia Resource” sip:[email protected];tag=as7529094e

To: sip:[email protected]:5070;transport=udp

Call-ID: [email protected]

Date: Sat, 16 May 2009 15:17:03 GMT

CSeq: 102 INVITE

Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 213.121.131.184:5070:
ACK sip:[email protected]:5070;transport=udp SIP/2.0

Via: SIP/2.0/UDP 81.138.84.236:5060;branch=z9hG4bK1bdff7dd;rport

From: “Julia Resource” sip:[email protected];tag=as7529094e

To: sip:[email protected]:5070;transport=udp

Contact: sip:[email protected]

Call-ID: [email protected]

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0


e[Khost81-138-84-236*CLI>
– SIP/103-097fd358 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [[email protected]:8] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mDIALSTATUS=CONGESTIONe[0;37;40m”) in new stack
– Executing [[email protected]:9] e[1;36;40mGosubIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?CONGESTION|1e[0;37;40m”) in new stack
– Executing [[email protected]:10] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?exit|returne[0;37;40m”) in new stack
– Executing [[email protected]:11] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mSV_DIALSTATUS=CONGESTIONe[0;37;40m”) in new stack
– Executing [[email protected]:12] e[1;36;40mGosubIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?docfu|1e[0;37;40m”) in new stack
– Executing [[email protected]:13] e[1;36;40mGosubIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?docfb|1e[0;37;40m”) in new stack
– Executing [[email protected]:14] e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mDIALSTATUS=CONGESTIONe[0;37;40m”) in new stack
– Executing [[email protected]:15] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mVoicemail is novme[0;37;40m”) in new stack
– Executing [[email protected]:16] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1?s-CONGESTION|1e[0;37;40m”) in new stack
– Goto (macro-exten-vm,s-CONGESTION,1)
– Executing [[email protected]:1] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mIVR_RETVM: IVR_CONTEXT: e[0;37;40m”) in new stack
– Executing [[email protected]:2] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m0?exit|1e[0;37;40m”) in new stack
– Executing [[email protected]:3] e[1;36;40mPlayTonese[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mcongestione[0;37;40m”) in new stack
– Executing [[email protected]:4] e[1;36;40mCongestione[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m10e[0;37;40m”) in new stack
== Spawn extension (macro-exten-vm, s-CONGESTION, 4) exited non-zero on ‘SIP/101-097fea60’ in macro ‘exten-vm’
== Spawn extension (from-internal, 103, 1) exited non-zero on ‘SIP/101-097fea60’
– Executing [[email protected]:1] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mhangupcalle[0;37;40m”) in new stack
– Executing [[email protected]:1] e[1;36;40mResetCDRe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40mvwe[0;37;40m”) in new stack

e[Khost81-138-84-236*CLI>
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
– Executing [[email protected]:2] e[1;36;40mNoCDRe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack
– Executing [[email protected]:3] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1?skiprge[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1?skipblkvme[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40m1?theende[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] e[1;36;40mHangupe[0;37;40m(“e[1;35;40mSIP/101-097fea60e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/101-097fea60’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/101-097fea60’

kismet,

I’ll take a shot here but I could be off base without more background being provided by you.

Two of the three extensions you show as registered are using port 5070 which is NOT a standard setup. SIP by default is connected to port 5060 unless you have gone in and re-configured things. If so please post what you have changed, etc. That way we can see what else you might have missed.

If not please post more information on your setup. See http://www.freepbx.org/forum/freepbx/installation/so-you-have-a-problem-and-want-help for what we are looking for.

Thanks Fskrotzki

I had problems getting 5060 past the router for some reason if I left it as defaut it came in on port 1025 never did get to the bottom of that, so I moved it to 5070

There a 3 cisco phones set up 2 x 7940 and 1 x 7912, its the 2 7940’s that register on port 5070, one of them works fine its the one set as extension 103 where outgoing is working ok but I cant get any calls to it.

Below are the configs from the 7940

The software is FreePBX 2.5.1.5 but I think that is all ok as the other 2 extensions work fine.
My next step was to wireshark and confirm if anything was being blocked

Thanks Paul

SIP Generic Configuration File For CISCO 7940 (start)

Voipfone

Image Version

image_version: “P0S3-08-9-00”

Proxy Server info

proxy1_address: "81.137.xx.xx"
proxy1_port:“5060”

proxy2_address: “81.137.xx.xx”"
proxy2_port:“5060”

proxy_register : 1

Outbound Proxy info

outbound_proxy: ""
outbound_proxy_port: “5060”

Emergency Proxy info

proxy_emergency: ""
proxy_emergency_port: “”

Backup Proxy info

proxy_backup: ""
proxy_backup_port: “”

NAT/Firewall Traversal

nat_enable: “1”

nat_received_processing: "1"
voip_control_port: "5070"
start_media_port: "16384"
end_media_port: “32766”

Time Zone phone will reside in

time_zone: GMT

date_format : D/M/Y

Setting for Message speeddial to UOne box

messages_uri: “199”

mwi_status : “0”

Phone Registration Expiration [1-3932100 sec] (Default - 3600)

timer_register_expires: “60”

Codec for media stream (g711ulaw (default), g711alaw, g729)

preferred_codec: “g711ulaw”

TOS bits in media stream [0-5] (Default - 5)

tos_media: “5”

Enable VAD (0-disable (default), 1-enable)

enable_vad: “0”

Allow for the bridge on a 3way call to join remaining parties upon hangup

cnf_join_enable: “1” ; 0-Disabled, 1-Enabled (default)

Allow Transfer to be completed while target phone is still ringing

semi_attended_transfer: “0” ; 0-Disabled, 1-Enabled (default)

Telnet Level (enable or disable the ability to telnet into this phone

telnet_level: “2” ; 0-Disabled (default), 1-Enabled, 2-Privileged

Inband DTMF Settings (0-disable, 1-enable (default))

dtmf_inband: “1”

Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )

dtmf_outofband: “avt”

DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)

dtmf_db_level: “3”

SIP Timers

timer_t1: “500” ; Default 500 msec

timer_t2: “4000” ; Default 4 sec

sip_retx: “10” ; Default 11

sip_invite_retx: “6” ; Default 7

timer_invite_expires: “180” ; Default 180 sec

TFTP Phone Specific Configuration File Directory

tftp_cfg_dir: “./”

Time Server

sntp_mode: “unicast”

sntp_server: “0.pool.ntp.org

time_zone: “GMT”

dst_offset: “1”

dst_start_month: “April”

dst_start_day: “”

dst_start_day_of_week: “Sun”

dst_start_week_of_month: “1”

dst_start_time: “02”

dst_stop_month: “Oct”

dst_stop_day: “”

dst_stop_day_of_week: “Sunday”

dst_stop_week_of_month: “8”

dst_stop_time: “2”

dst_auto_adjust: “1”

Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)

dnd_control: “2” ; Default 0 (Do Not Disturb feature is off)

Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)

callerid_blocking: “0” ; Default 0 (Disable sending all calls as anonymous)

Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)

anonymous_call_block: “0” ; Default 0 (Disable blocking of anonymous calls)

Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)

call_waiting: “1” ; Default 1 (Call Waiting enabled)

DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)

dtmf_avt_payload: “101” ; Default 100

XML file that specifies the dialplan desired

dial_template: “dialplan”

Network Media Type (auto, full100, full10, half100, half10)

network_media_type: “auto”

#Autocompletion During Dial (0-off, 1-on [default])

autocomplete: “0”

#Time Format (0-12hr, 1-24hr [default])

time_format_24hr: “1”

URL for external Phone Services

services_url: “http://192.168.1.21/phone/news.htm” ; News Feed Services

services_url: “http://voipfone.co.uk

URL for external Directory location

directory_url: “http://192.168.1.51/voip/directory.html” ; Phone Directory

URL for branding logo

logo_url: “http://192.168.1.51/voip/map2.bmp

logo_url: “10-20logo.bmp”

Remote Party ID

remote_party_id: 1 ; 0-Disabled (default), 1-Enabled

SIP Generic Configuration File For CISCO 7940 (end)

This is the mac address file for the phone specifics

SIP Configuration custom File

Line 1 appearance

line1_name: “103”

line1_shortname : “Resource 103”

Line 1 Registration Authentication

line1_authname: “103”

Line 1 Registration Password

line1_password: “secret”

line1_contact : “103”

####### New Parameters added in Release 2.0 #######

All user_parameters have been removed

Phone Label (Text desired to be displayed in upper right corner)

phone_label: “” ; Has no effect on SIP messaging

Line 1 Display Name (Display name to use for SIP messaging)

line1_displayname: “”

Line 2 Display Name (Display name to use for SIP messaging)

#line2_displayname: “”

####### New Parameters added in Release 3.0 ######

Phone Prompt (The prompt that will be displayed on console and telnet)

phone_prompt: “SIP Phone 1” ; Limited to 15 characters (Default - SIP Phone)

Phone Password (Password to be used for console or telnet login)

phone_password: “xxxxxxx” ; Limited to 31 characters (Default - cisco)

User classifcation used when Registering [ none(default), phone, ip ]

user_info: none

ok I’m 98% sure this is all firewall issues. Since the phones you are connecting to are external and you have stated that you have firewall problems with 5060 I’m guessing that you are not configured properly.

So details on the firewall to start please.

You do realize that you need to open up UDP port 5060 and UDP 10000 to 20000 and forward all of them to the server. At the same time you need to create and add the needed lines in EITHER sip_nat.conf (old format) or sip_general_custom.conf (new format) and then reload. For details please see: http://freepbx.org/configuration_files

At the same time if there are firewalls at the remote locations they can be a part of the problem. Some older firewalls may need to have port 5060 forwarded to the phone.

Thanks Again
The server is on a public IP, only the phones use nat. mods to sip_general_custom.conf only seems to be needed when using the server behind nat, is this incorrect ??

It is firewalled but all the ports you specify are allowed.

The phone in question has been put in the DMZ of the router so realistically it should have no problems, I think I will go to the site and use wireshark to confirm if anything is being blocked.

Thanks for your suggestions
Paul

Not Managed to get to site yet but have done a bit more diags with the phone, these are the logs. it looks like the call is getting to it so the port forwarding seems to be working
this is extension 111 calling 112 again failing

couple of errors I can see are

Unknown address in Request URI
SIP/2.0 404 Not Found

SIP Phone 1> [09:51:05:166921] SIPProcessUDPMessage: recv UDP message from <81.
37.84.237>:<50195>, length=<921>, message=
[09:51:05:166921] INVITE sip:[email protected]:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 80.137.84.237:5060;branch=z9hG4bK4e39e9bf;rport
From: “Julia” sip:[email protected];tag=as0b68ab85
To: sip:[email protected]:5061;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 19 May 2009 08:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 363

v=0
o=root 2636 2636 IN IP4 80.137.84.237
s=session
c=IN IP4 80.137.84.237
b=CT:384
t=0 0
m=audio 13456 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 13798 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv
[09:51:05:166925] Unknown address in Request URI
[09:51:05:166926] sipSPICheckRequest: Request URI Not Found
[09:51:05:166926] SIPTaskProcessSIPMessage: Error: sipSPICheckRequest() returne
error.
[09:51:05:166928] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, met
od=INVITE, to_tag=<>)
[09:51:05:166929] sipTransportSendMessage: Opened a one-time UDP send channel t
server <80.137.84.237>:<5060>, handle = 8 local port= 5061
[09:51:05:166930] sipTransportSendMessage:Sent SIP message to <80.137.84.237>:<
060>, handle=<8>, length=<326>, message=
[09:51:05:166930] SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 80.137.84.237:5060;branch=z9hG4bK4e39e9bf;rport
From: “Julia” sip:[email protected];tag=as0b68ab85
To: sip:[email protected]:5061;transport=udp
Call-ID: [email protected]
Date: Tue, 19 May 2009 08:51:05 GMT
CSeq: 102 INVITE
Content-Length: 0

[09:51:05:166932] sipTransportSendMessage: Closed a one-time UDP send channel h
ndle = 8
[09:51:05:166938] SIPProcessUDPMessage: recv UDP message from <80.137.84.237>:<
0195>, length=<393>, message=
[09:51:05:166939] ACK sip:[email protected]:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 80.137.84.237:5060;branch=z9hG4bK4e39e9bf;rport
From: “Julia” sip:[email protected];tag=as0b68ab85
To: sip:[email protected]:5061;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0