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pbx support Newbie

pbx support

Posted
Posted

It's been a while since we released the last bug fix version(2011-4.2.0.3981) for snomONE. In the meanwhile, we heard the feedback from you in the form of forum requests, support tickets, etc. We tried to resolve most of the known issues during this time and as a result of this is the latest bug fix release 2011-4.2.1.4025.

 

For details such as release notes and download links please refer the below link

 

http://wiki.snomone.com/index.php?title=Release_notes

 

Thank you all!!!

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pbx support Newbie

pbx support

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  • Author
Posted

Dear,

 

Thanks for the update.

After update I can no longer start the MacOSX snomOne Free version. Key is not correct for this version !

But on the wiki it states it is for all versions.

 

Any idea what can be wrong

 

regards

Fonny

There should not be anything special here. We will double check.

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Vodia support Proficient

Vodia support

Posted
Posted

V, if you want to update the snomONE plus here is how.

 

 

Upgrading/Updating the snomONE

 

Once the snomONE is installed and running succefully, the upgrade to a newer version is very simple.

 

Download the new version to snomONE directory

Make the file executable using chmod a+x <new file> command.

Stop the snomONE using service snomONE stop command.

Point the link to new binary using ln -sf <new file> snomONE-ctrl command.

Start the snomONE using service snomONE start command.

This procedure has to be followed anytime you want to upgrade (or downgrade) the snomONE.

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p800aul Newbie

p800aul

Posted
Posted

It's been a while since we released the last bug fix version(2011-4.2.0.3981) for snomONE. In the meanwhile, we heard the feedback from you in the form of forum requests, support tickets, etc. We tried to resolve most of the known issues during this time and as a result of this is the latest bug fix release 2011-4.2.1.4025.

 

For details such as release notes and download links please refer the below link

 

http://wiki.snomone.com/index.php?title=Release_notes

 

Thank you all!!!

 

Any reason why the patton trunk will not dial out using this the sipgate still works.

 

So previous version 2011-4.2.0.3981 worked incoming - outgoing sipgate and worked incoming - outgoing patton 2xfxo

 

Upgrade to 2011-4.2.1.4025, sipgate incoming - outgoing, patton incoming only - no outgoing (engaged tone).

 

Downgrade back to 2011-4.2.0.3981 everything fine again.

 

Regards

 

Paul

 

patton trunk set up:

 

# Trunk 5 in domain localhost

Name: Patton

Type: register

To: sip

RegPass: ********

Direction:

Disabled: false

Global: false

Display:

RegAccount:

RegRegistrar: 192.168.1.200

RegKeep:

RegUser:

Icid:

Require:

OutboundProxy: 192.168.1.200

Ani:

DialExtension: 72

Prefix:

Trusted: false

AcceptRedirect: false

RfcRtp: false

Analog: false

SendEmail:

UseUuid: false

Ring180: false

Failover: only_5xx

Privacy: false

Glob:

RequestTimeout:

Codecs:

CodecLock: true

Expires: 3600

FromUser:

Tel: true

TranscodeDtmf: false

AssociatedAddresses:

InterOffice: false

DialPlan:

Colines:

DialogPermission:

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p800aul Newbie

p800aul

Posted
Posted

There you go

 

Thanks

 

 

.3981

 

 

[6] 2011/07/01 18:09:22:

Received bindRequest for user localhost\48



[5] 2011/07/01 18:09:25:

SIP Rx tls:192.168.1.8:2778:



INVITE sip:819161@localhost;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport
From: "Study" <sip:48@localhost>;tag=qzv6db7x6u
To: <sip:819161@localhost;user=phone>
Call-ID: 3c641d1fe2f7-4xrvrha9is0i
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:48@192.168.1.8:2778;transport=tls;line=mjvwc7ij>;reg-id=1
X-Serialnumber: 00041336B86D
P-Key-Flags: keys="3"
User-Agent: snom300/8.4.31
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 520

v=0
o=root 391218360 391218360 IN IP4 192.168.1.8
s=call
c=IN IP4 192.168.1.8
t=0 0
m=audio 56900 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:AbJQ3lgRvtJ7BbbTxRK15Rg3nXBsgROXGul4+G7J
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv




[8] 2011/07/01 18:09:25:

Packet authenticated by transport layer



[9] 2011/07/01 18:09:25:

UDP: Opening socket on 0.0.0.0:58970



[9] 2011/07/01 18:09:25:

UDP: Opening socket on 0.0.0.0:58971



[8] 2011/07/01 18:09:25:

Could not find a trunk (2 trunks)



[5] 2011/07/01 18:09:25:

SIP Rx tls:192.168.1.8:2778:



INVITE sip:819161@localhost;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport
From: "Study" <sip:48@localhost>;tag=qzv6db7x6u
To: <sip:819161@localhost;user=phone>
Call-ID: 3c641d1fe2f7-4xrvrha9is0i
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:48@192.168.1.8:2778;transport=tls;line=mjvwc7ij>;reg-id=1
X-Serialnumber: 00041336B86D
P-Key-Flags: keys="3"
User-Agent: snom300/8.4.31
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 520

v=0
o=root 391218360 391218360 IN IP4 192.168.1.8
s=call
c=IN IP4 192.168.1.8
t=0 0
m=audio 56900 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:AbJQ3lgRvtJ7BbbTxRK15Rg3nXBsgROXGul4+G7J
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv




[9] 2011/07/01 18:09:25:

Using outbound proxy sip:192.168.1.8:2778;transport=tls because of flow-label



[9] 2011/07/01 18:09:25:

Last message repeated 3 times



[6] 2011/07/01 18:09:25:

Received bindRequest for user localhost\48



[5] 2011/07/01 18:09:25:

SIP Tx tls:192.168.1.8:2778:



SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport=2778
From: "Study" <sip:48@localhost>;tag=qzv6db7x6u
To: <sip:819161@localhost;user=phone>;tag=ebe544b72c
Call-ID: 3c641d1fe2f7-4xrvrha9is0i
CSeq: 1 INVITE
Content-Length: 0





[7] 2011/07/01 18:09:25:

Set packet length to 20



[6] 2011/07/01 18:09:25:

Sending RTP for 3c641d1fe2f7-4xrvrha9is0i to 192.168.1.8:56900, codec not set yet



[8] 2011/07/01 18:09:25:

Call from an user 48



[8] 2011/07/01 18:09:25:

To is <sip:819161@localhost;user=phone>, user 0, domain 1



[8] 2011/07/01 18:09:25:

From user 48



[8] 2011/07/01 18:09:25:

Set the To domain based on From user 48@localhost



[8] 2011/07/01 18:09:25:

Call state for call object 13: idle



[7] 2011/07/01 18:09:25:

set_codecs: for 3c641d1fe2f7-4xrvrha9is0i codecs "", codec_preference count 6



[9] 2011/07/01 18:09:25:

Dialplan: Evaluating !^(999)@.*!sip:\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:09:25:

Dialplan: Evaluating !^0800([0-9]*)@.*!sip:0800\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:09:25:

Dialplan: Evaluating !^00([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:09:25:

Dialplan: Evaluating !^07([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:09:25:

Dialplan: Evaluating !^907([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:09:25:

Dialplan: Evaluating !^900([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:09:25:

Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 819161@localhost



[5] 2011/07/01 18:09:25:

Dialplan "Standard Dialplan": Match 819161@localhost to <sip:819161@192.168.1.200;user=phone> on trunk Patton



[8] 2011/07/01 18:09:25:

Play audio_moh/noise.wav



[9] 2011/07/01 18:09:25:

UDP: Opening socket on 0.0.0.0:59340



[9] 2011/07/01 18:09:25:

UDP: Opening socket on 0.0.0.0:59341



[7] 2011/07/01 18:09:25:

set_codecs: for 83572110@pbx codecs "", codec_preference count 6



[9] 2011/07/01 18:09:25:

update_codecs for 83572110@pbx: adding codec pcmu/8000 to available list



[9] 2011/07/01 18:09:25:

update_codecs for 83572110@pbx: adding codec pcma/8000 to available list



[9] 2011/07/01 18:09:25:

update_codecs for 83572110@pbx: adding codec g722/8000 to available list



[9] 2011/07/01 18:09:25:

update_codecs for 83572110@pbx: adding codec g726-32/8000 to available list



[9] 2011/07/01 18:09:25:

update_codecs for 83572110@pbx: adding codec gsm/8000 to available list



[9] 2011/07/01 18:09:25:

update_codecs for 83572110@pbx: codec_preference size 6, available codecs size 6



[9] 2011/07/01 18:09:25:

Resolve 12472: url sip:192.168.1.200



[9] 2011/07/01 18:09:25:

Resolve 12472: udp 192.168.1.200 5060



[5] 2011/07/01 18:09:25:

SIP Tx udp:192.168.1.200:5060:



INVITE sip:819161@192.168.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-72902ae4db65e28d50e1980ed6df68bf;rport
From: "Study" <sip:01246819161@localhost;user=phone>;tag=45409
To: <sip:819161@192.168.1.200;user=phone>
Call-ID: 83572110@pbx
CSeq: 900 INVITE
Max-Forwards: 70
Contact: <sip:01246819161@192.168.1.13:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Type: application/sdp
Content-Length: 327

v=0
o=- 41145 41145 IN IP4 192.168.1.13
s=-
c=IN IP4 192.168.1.13
t=0 0
m=audio 59340 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

 

4025

 

 

[6] 2011/07/01 18:16:31:

Received bindRequest for user localhost\48



[6] 2011/07/01 18:16:33:

Last message repeated 2 times



[7] 2011/07/01 18:16:33:

SIP Rx tls:192.168.1.8:2782:



INVITE sip:819161@localhost;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport
From: "Study" <sip:48@localhost>;tag=xy3mypr1cv
To: <sip:819161@localhost;user=phone>
Call-ID: 3c641eccef30-an7w0fgu4eg2
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:48@192.168.1.8:2782;transport=tls;line=mjvwc7ij>;reg-id=1
X-Serialnumber: 00041336B86D
P-Key-Flags: keys="3"
User-Agent: snom300/8.4.31
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 522

v=0
o=root 1034786031 1034786031 IN IP4 192.168.1.8
s=call
c=IN IP4 192.168.1.8
t=0 0
m=audio 54922 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XhQqTHLltypTJWC5vDrHpGfZkxH45okk1VH+jdSi
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv




[8] 2011/07/01 18:16:33:

Packet authenticated by transport layer



[9] 2011/07/01 18:16:33:

UDP: Opening socket on 0.0.0.0:60914



[9] 2011/07/01 18:16:33:

UDP: Opening socket on 0.0.0.0:60915



[8] 2011/07/01 18:16:33:

Could not find a trunk (2 trunks)



[9] 2011/07/01 18:16:33:

Using outbound proxy sip:192.168.1.8:2782;transport=tls because of flow-label



[9] 2011/07/01 18:16:33:

Last message repeated 3 times



[7] 2011/07/01 18:16:33:

SIP Tx tls:192.168.1.8:2782:



SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport=2782
From: "Study" <sip:48@localhost>;tag=xy3mypr1cv
To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d
Call-ID: 3c641eccef30-an7w0fgu4eg2
CSeq: 1 INVITE
Content-Length: 0





[7] 2011/07/01 18:16:33:

Set packet length to 20



[6] 2011/07/01 18:16:33:

Sending RTP for 3c641eccef30-an7w0fgu4eg2 to 192.168.1.8:54922, codec not set yet



[8] 2011/07/01 18:16:33:

Incoming call: Request URI sip:819161@localhost;user=phone, To is <sip:819161@localhost;user=phone>



[8] 2011/07/01 18:16:33:

Call from an user 48



[8] 2011/07/01 18:16:33:

To is <sip:819161@localhost;user=phone>, user 0, domain 1



[8] 2011/07/01 18:16:33:

From user 48



[8] 2011/07/01 18:16:33:

Set the To domain based on From user 48@localhost



[8] 2011/07/01 18:16:33:

Call state for call object 1: idle



[7] 2011/07/01 18:16:33:

set_codecs: for 3c641eccef30-an7w0fgu4eg2 codecs "", codec_preference count 6



[9] 2011/07/01 18:16:33:

Dialplan: Evaluating !^(999)@.*!sip:\1@\r;user=phone!i against 819161@localhost



[6] 2011/07/01 18:16:33:

The registration type trunk Patton is not registered. Skipping it...



[9] 2011/07/01 18:16:33:

Dialplan: Evaluating !^0800([0-9]*)@.*!sip:0800\1@\r;user=phone!i against 819161@localhost



[6] 2011/07/01 18:16:33:

The registration type trunk Patton is not registered. Skipping it...



[9] 2011/07/01 18:16:33:

Dialplan: Evaluating !^00([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:16:33:

Dialplan: Evaluating !^07([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost



[9] 2011/07/01 18:16:33:

Dialplan: Evaluating !^907([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost



[6] 2011/07/01 18:16:33:

The registration type trunk Patton is not registered. Skipping it...



[9] 2011/07/01 18:16:33:

Dialplan: Evaluating !^900([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost



[6] 2011/07/01 18:16:33:

The registration type trunk Patton is not registered. Skipping it...



[9] 2011/07/01 18:16:33:

Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 819161@localhost



[6] 2011/07/01 18:16:33:

The registration type trunk Patton is not registered. Skipping it...



[8] 2011/07/01 18:16:33:

call port 0: state code from 0 to 404



[7] 2011/07/01 18:16:33:

Set packet length to 20



[7] 2011/07/01 18:16:33:

SIP Tx tls:192.168.1.8:2782:



SIP/2.0 404 Not Found
Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport=2782
From: "Study" <sip:48@localhost>;tag=xy3mypr1cv
To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d
Call-ID: 3c641eccef30-an7w0fgu4eg2
CSeq: 1 INVITE
Contact: <sip:48@192.168.1.13:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.1.4025
Content-Length: 0





[6] 2011/07/01 18:16:33:

Received searchRequest, equalityMatch (description=telephoneNumber, value=819161)



[7] 2011/07/01 18:16:33:

SIP Rx tls:192.168.1.8:2782:



ACK sip:819161@localhost;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport
From: "Study" <sip:48@localhost>;tag=xy3mypr1cv
To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d
Call-ID: 3c641eccef30-an7w0fgu4eg2
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:48@192.168.1.8:2782;transport=tls;line=mjvwc7ij>;reg-id=1
Proxy-Require: buttons
Content-Length: 0

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tommymtl Newbie

tommymtl

Posted
Posted

A simple question

 

for PBXnSIP user ... 4.2.0.3981(win32) was 9,20 MB (9 649 152 bytes)

and new version on PBXnSIP 4.2.1.4025 (Win32) is now 7,22 MB (7 572 480 bytes)...

Over 2 mb smaller...

 

what was removed?

what are we missing?

 

Are we loosing something or does the update should work seamless? with all complex elements, we are getting nervous ...

 

thanks

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pbx support Newbie

pbx support

Posted
  • Author
Posted

Oops, missed that totally.

 

The issue is that on the PBX, "Patton" trunk type is set to "SIP Registration", but the trunk is not registered. In .4025, we skip the unregistered trunks for outbound calls. You can do either - make sure that the trunk is registered OR change the trunk type to "SIP Gateway" to avoid the issue.

 

The registration type trunk Patton is not registered. Skipping it...

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p800aul Newbie

p800aul

Posted
Posted

Oops, missed that totally.

 

The issue is that on the PBX, "Patton" trunk type is set to "SIP Registration", but the trunk is not registered. In .4025, we skip the unregistered trunks for outbound calls. You can do either - make sure that the trunk is registered OR change the trunk type to "SIP Gateway" to avoid the issue.

 

The registration type trunk Patton is not registered. Skipping it...

 

Thanks

 

That seems to have fixed it

 

Regards

 

Paul

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hmcap Newbie

hmcap

Posted
Posted

I'll post my question here and also in 'general', but I pretty desperate and don't know where to start. It happens after upgrading to 4.2.1.4025. First I had some problems with the caller-ID and now all my inbound trunks are not working. I've a DID via DIDWW, which is mapped to a voipcheap-trunk with a failover to a voipbuster-trunk. Since two days, it is not possible anymore to call our office. A long silence is heard. I than down=graded to x.x.x.3981 and it looks like the snom one was working again. Just for a few minutes and the office was not reacheable again. How to solve this? I don't know where to start looking, but it has to work.

 

Regards,

 

Harry

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  • 4 weeks later...
andrewgroup Newbie

andrewgroup

Posted
Posted

If or when will SnomOne tightly integrate with Snom Phones. By this I mean control the many settings in the phones from a WEB interface on the PBX.

We are fine with PNP, and buttons, but being able to set global Snom Settings in the PBX and control on a EXT by EXT basis the many other settings that you commonly will need to adjust. Might this be on the Road Map?

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  • 3 weeks later...
Ganesh Newbie

Ganesh

Posted
Posted

Hello,

 

We are using Cent OS and upgraded the pbxnsip to version 4.2.1.4025. After upgrade while I try to start the service I see the error as "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR"

 

The start up script file works fine with the older version and not with version 4025. Can you please tell me what changes we need to make in start up file? I have checked the installation directory. It is as per the path in the start up file.

 

Regards

Ganesh

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Ganesh Newbie

Ganesh

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Posted

Hello,

 

We are using Cent OS and upgraded the pbxnsip to version 4.2.1.4025. After upgrade while I try to start the service I see the error as "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR"

 

The start up script file works fine with the older version and not with version 4025. Can you please tell me what changes we need to make in start up file? I have checked the installation directory. It is as per the path in the start up file.

 

Regards

Ganesh

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Ganesh Newbie

Ganesh

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Posted

This is the startup file that we are using for CentOS and this is working with older versions and not working with version 4025. The error displayed while starting the service is "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR"

Let me know the changes needed in the file to run version 4025.

 

#!/bin/bash

#

# Init file for pbxnsip PBX

#

# Copyright © 2006 pbxnsip Inc., USA

#

# chkconfig: 2345 20 80

# description: SIP-based PBX

#

# processname: pbxctrl

# pidfile: /var/run/pbxctrl.pid

# source function library

. /etc/rc.d/init.d/functions

RETVAL=0

# Installation location

INSTALLDIR=/usr/local/pbxnsip

PBX=$INSTALLDIR/pbxctrl

start()

{

echo -n "Starting PBX:"

$PBX --dir $INSTALLDIR

echo

RETVAL=1

}

stop()

{

echo -n "Stopping PBX:"

killproc $PBX -TERM

echo

RETVAL=1

}

case "$1" in

start)

start

;;

stop)

stop

;;

restart)

stop

start

;;

status)

status $PBX

RETVAL=$?

;;

*)

echo $"Usage: $0 {start|stop|restart|status}"

RETVAL=1

esac

exit $RETVAL

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pbx support Newbie

pbx support

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There are no changes required for the new version. All you have to do is to download the new version from pbxnsip download page and follow http://wiki.snomone.com/index.php?title=Upgrades#General_manual_upgrade_guidelines_for_Linux_based_systems. This one is written with snomONE in mind. But you can substitute pbxctrl instead and have it working.

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