Jump to content

Recommended Posts

We released 2.1.7, release notes as usual to be found on http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.7. This update is recommended for users that are running 2.1.6, it only fixes problems that were found in 2.1.6 and does not have any new features. For the CS410 we recommend to use the 3.0 build, as it includes important flags for the FXO subsystem.

Link to post
Share on other sites
We released 2.1.7, release notes as usual to be found on http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.7. This update is recommended for users that are running 2.1.6, it only fixes problems that were found in 2.1.6 and does not have any new features. For the CS410 we recommend to use the 3.0 build, as it includes important flags for the FXO subsystem.

 

I noticed that the CS410 update to version 3.0.0.2899 still points to the 2.1.7 release notes. What are the differences between the 2.1.7 software release and the major version number bump to 3.x for the CS410? We have several CS410s deployed at client sites and we want to make sure that all our bases are covered before we push the update to them.

 

--

Tim Donahue

Link to post
Share on other sites

That is a tricky topic. Of course the intention was to keep the CS410 on the same path as all other software - but we simply have the problem that we need the settings for CPC duration, polarity change and busy detection which does not exist in 2.1. There are no "dangerous" changes in the 3.0 version yet, IMHO so far is it safe to use it. The release notes include everything between 2.1.6 and 2.1.7.

Link to post
Share on other sites

I am running 2.1.8 and the PBXnSIP is now offering call camp when the extension does not answer, and rolls to voicemail. It is also offering call camp to external callers. Is there a way to make it only offer when the call comes from the inside, and the extension is in use?

Link to post
Share on other sites
I am running 2.1.8 and the PBXnSIP is now offering call camp when the extension does not answer, and rolls to voicemail. It is also offering call camp to external callers. Is there a way to make it only offer when the call comes from the inside, and the extension is in use?

 

You mean from any external number? Or just numbers that are listed as cell phone in an extension?

Link to post
Share on other sites

You are correct, it only offered camp on when calling from a cell phone associated with an internal account. Now that I realize that, I can accept it, however I cannot accept that it offers when there is no answer.

Link to post
Share on other sites
You are correct, it only offered camp on when calling from a cell phone associated with an internal account. Now that I realize that, I can accept it, however I cannot accept that it offers when there is no answer.

 

Why not for "no answer"? I think it is nice if someone is out for lunch, you want to talk to him and get a call back after he is back from lunch and finishes the first call.

Link to post
Share on other sites

Version 1.xx had issues with devices behind NATs. Incoming calls were blocked by NAT and a caller hears silence before the voicemail answers.

 

version 2.xx (new install) does not have this issue and we tested it with the device behind many NAT devices and it works.

 

The problem we are facing is that we have several PBXes that started as 1.xxx and we upgraded to 2.xxx. The problem with the NAT remains. The same problem occurs for Susee and Debian flavours of the PBX. We are now using 2.1.7.2461 (Linux).

 

A call from an outside number 778-893-9348 (cell) to a voice over IP line 604-637-0912. The end user device is registered with the PBX from behind a NAT. When the call comes in, the caller hears an extended silence and then get forwarded to the voicemail. The device does not see the call coming in. (Please note that when we use a new version 2.xx install, it works and the call rings on the end user device correctly).

 

This is trace from a 2.1.7.2461 version of software that has been upgraded (multiple upgrades) from a 1.xxx version:

 

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 208.68.18.226;branch=z9hG4bK5292.43508421.1

Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK5292.7a9b8681.0

Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-efaaaf4788145af9e2c8baffc0c7212e-159.18.161.101-1

Record-Route: <sip:208.68.18.226;lr=on;ftag=159.18.161.101+1+4fd41c+4a38210d>

Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on>

From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00;tag=159.18.161.101+1+4fd41c+4a38210d

To: <sip:6046370912@208.68.18.226>;tag=030856d931

Call-ID: 66876E82@159.18.161.101

CSeq: 154088517 INVITE

Contact: <sip:6046370912@208.68.18.228:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: netfone-PBX/2.1.7.2461

Content-Type: application/sdp

Content-Length: 262

 

v=0

o=- 1325934170 1325934170 IN IP4 208.68.18.228

s=-

c=IN IP4 208.68.18.228

t=0 0

m=audio 5782 RTP/AVP 18 0 101

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[9] 2008/04/25 07:53:46: SIP Rx udp:159.18.161.67:5060:

ACK sip:6046370912@208.68.18.228:5060;transport=udp SIP/2.0

Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on>

Via: SIP/2.0/UDP 159.18.161.67;branch=0

Via: SIP/2.0/UDP 159.18.161.101:5060;branch=z9hG4bK-37fd8013b7ae6f3f13c353abfbc326eb-159.18.161.101-1

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info

Max-Forwards: 69

Call-ID: 66876E82@159.18.161.101

From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=159.18.161.101+1+4fd41c+4a38210d;isup-oli=00

To: <sip:6046370912@208.68.18.226>;tag=030856d931

CSeq: 154088517 ACK

Contact: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00

Organization: MetaSwitch

Content-Length: 0

 

 

[9] 2008/04/25 07:53:46: SIP Tr udp:207.6.229.160:60778:

CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport

From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482

To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca>

Call-ID: 5dc9a996@pbx

CSeq: 27432 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[9] 2008/04/25 07:53:51: Last message repeated 3 times

 

[9] 2008/04/25 07:53:51: SIP Rx udp:159.18.161.67:5060:

BYE sip:6046370912@208.68.18.228:5060;transport=udp SIP/2.0

Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on>

Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK3392.ee170516.0

Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-64fdd954f60e668f7627d251a924542b-159.18.161.101-1

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info

Max-Forwards: 69

Call-ID: 66876E82@159.18.161.101

From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=159.18.161.101+1+4fd41c+4a38210d;isup-oli=00

To: <sip:6046370912@208.68.18.226>;tag=030856d931

CSeq: 154088518 BYE

Organization: MetaSwitch

Supported: 100rel

Content-Length: 0

 

 

[9] 2008/04/25 07:53:51: Resolve 317750: aaaa udp 159.18.161.67 5060

[9] 2008/04/25 07:53:51: Resolve 317750: a udp 159.18.161.67 5060

[9] 2008/04/25 07:53:51: Resolve 317750: udp 159.18.161.67 5060

[9] 2008/04/25 07:53:51: SIP Tx udp:159.18.161.67:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK3392.ee170516.0

Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-64fdd954f60e668f7627d251a924542b-159.18.161.101-1

Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on>

From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00;tag=159.18.161.101+1+4fd41c+4a38210d

To: <sip:6046370912@208.68.18.226>;tag=030856d931

Call-ID: 66876E82@159.18.161.101

CSeq: 154088518 BYE

Contact: <sip:6046370912@208.68.18.228:5060;transport=udp>

User-Agent: netfone-PBX/2.1.7.2461

RTP-RxStat: Dur=25,Pkt=259,Oct=8288,Underun=0

RTP-TxStat: Dur=5,Pkt=259,Oct=8288

Content-Length: 0

 

 

[7] 2008/04/25 07:53:51: Other Ports: 5

[7] 2008/04/25 07:53:51: Call Port: 020ec8ff@pbx#1390526460

[7] 2008/04/25 07:53:51: Call Port: 5dc9a996@pbx#1253982482

[7] 2008/04/25 07:53:51: Call Port: ab17b219@pbx#327669878

[7] 2008/04/25 07:53:51: Call Port: b9b50168-472333bb@24.87.11.54#fc2e71a790

[7] 2008/04/25 07:53:51: Call Port: fd537cda-bd2baccc@10.0.0.3#35a4289630

[9] 2008/04/25 07:53:51: Using outbound proxy sip:207.6.229.160:60778;transport=udp because of flow-label

[9] 2008/04/25 07:53:51: Resolve 317751: url sip:207.6.229.160:60778;transport=udp

[9] 2008/04/25 07:53:51: Resolve 317751: a udp 207.6.229.160 60778

[9] 2008/04/25 07:53:51: Resolve 317751: udp 207.6.229.160 60778

[9] 2008/04/25 07:53:51: SOAP: Store CDR in http://208.68.18.230/call_logging/call_log.2.0.2.php

<env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://soap.com/pbx"><env:Body><sns:CDR><CallID>66876E82@159.18.161.101#030856d931</CallID><Type>mailbox</Type><Domain>office2.voipportal.ca</Domain><From>RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00</From><To>"Netfone Telesales" <sip:6046370912@office2.voipportal.ca></To><ToUser>6046370912@office2.voipportal.ca</ToUser><FromTrunk>van1_ser1</FromTrunk><TimeStart>1209135206</TimeStart><TimeEnd>1209135231</TimeEnd><StatisticsForward>0</StatisticsForward></sns:CDR></env:Body></env:Envelope>

 

[9] 2008/04/25 07:53:51: SIP Tx udp:207.6.229.160:60778:

NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport

From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c

To: Netfone Telesales <sip:6046370912@office2.voipportal.ca>

Call-ID: go2zi5t9@pbx

CSeq: 31263 NOTIFY

Max-Forwards: 70

Contact: <sip:208.68.18.228:5060;transport=udp>

Event: message-summary

Subscription-State: terminated;reason=noresource

Content-Type: application/simple-message-summary

Content-Length: 103

 

Messages-Waiting: no

Message-Account: sip:6046370912@office2.voipportal.ca

Voice-Message: 0/0 (0/0)

 

 

[9] 2008/04/25 07:53:52: SIP Tr udp:207.6.229.160:60778:

NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport

From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c

To: Netfone Telesales <sip:6046370912@office2.voipportal.ca>

Call-ID: go2zi5t9@pbx

CSeq: 31263 NOTIFY

Max-Forwards: 70

Contact: <sip:208.68.18.228:5060;transport=udp>

Event: message-summary

Subscription-State: terminated;reason=noresource

Content-Type: application/simple-message-summary

Content-Length: 103

 

Messages-Waiting: no

Message-Account: sip:6046370912@office2.voipportal.ca

Voice-Message: 0/0 (0/0)

 

 

[9] 2008/04/25 07:53:52: Message repetition, packet dropped

[8] 2008/04/25 07:53:52: SMTP: Connect to 64.40.101.128:25

[9] 2008/04/25 07:53:53: SIP Tr udp:207.6.229.160:60778:

NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport

From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c

To: Netfone Telesales <sip:6046370912@office2.voipportal.ca>

Call-ID: go2zi5t9@pbx

CSeq: 31263 NOTIFY

Max-Forwards: 70

Contact: <sip:208.68.18.228:5060;transport=udp>

Event: message-summary

Subscription-State: terminated;reason=noresource

Content-Type: application/simple-message-summary

Content-Length: 103

 

Messages-Waiting: no

Message-Account: sip:6046370912@office2.voipportal.ca

Voice-Message: 0/0 (0/0)

 

 

[9] 2008/04/25 07:53:53: Message repetition, packet dropped

[9] 2008/04/25 07:53:53: SIP Tr udp:207.6.229.160:60778:

CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport

From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482

To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca>

Call-ID: 5dc9a996@pbx

CSeq: 27432 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[5] 2008/04/25 07:53:54: Call 5dc9a996@pbx#1253982482: Last request not finished

[9] 2008/04/25 07:53:54: Resolve 317753: udp 207.6.229.160 60778

[9] 2008/04/25 07:53:54: SIP Tx udp:207.6.229.160:60778:

CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport

From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482

To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca>

Call-ID: 5dc9a996@pbx

CSeq: 27432 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[8] 2008/04/25 07:53:54: Hangup: Call 5dc9a996@pbx#1253982482 not found

[9] 2008/04/25 07:53:54: SIP Tr udp:207.6.229.160:60778:

CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport

From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482

To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca>

Call-ID: 5dc9a996@pbx

CSeq: 27432 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[9] 2008/04/25 07:53:55: SIP Tr udp:207.6.229.160:60778:

NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport

From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c

To: Netfone Telesales <sip:6046370912@office2.voipportal.ca>

Call-ID: go2zi5t9@pbx

CSeq: 31263 NOTIFY

Max-Forwards: 70

Contact: <sip:208.68.18.228:5060;transport=udp>

Event: message-summary

Subscription-State: terminated;reason=noresource

Content-Type: application/simple-message-summary

Content-Length: 103

 

Messages-Waiting: no

Message-Account: sip:6046370912@office2.voipportal.ca

Voice-Message: 0/0 (0/0)

 

 

 

[9] 2008/04/25 07:53:55: Message repetition, packet dropped

[9] 2008/04/25 07:53:55: SIP Tr udp:207.6.229.160:60778:

CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport

From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482

To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca>

Call-ID: 5dc9a996@pbx

CSeq: 27432 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[8] 2008/04/25 07:53:55: SMTP: Received 220 mail.netfone.ca ESMTP

 

[8] 2008/04/25 07:53:55: SMTP: Received 250-mail.netfone.ca

250-AUTH=LOGIN

250-AUTH LOGIN

250-PIPELINING

250 8BITMIME

 

[8] 2008/04/25 07:53:55: SMTP: Received 334 VXNlcm5hbWU6

 

[8] 2008/04/25 07:53:56: SMTP: Received 334 UGFzc3dvcmQ6

 

[8] 2008/04/25 07:53:56: SMTP: Received 235 go ahead

 

[8] 2008/04/25 07:53:56: SMTP: Received 221 mail.netfone.ca

 

[8] 2008/04/25 07:53:56: Sucessfully sent email to <rafehh@yahoo.com>

[9] 2008/04/25 07:53:57: SIP Tr udp:207.6.229.160:60778:

CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport

From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482

To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca>

Call-ID: 5dc9a996@pbx

CSeq: 27432 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[9] 2008/04/25 07:53:59: SIP Tr udp:207.6.229.160:60778:

NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport

From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c

To: Netfone Telesales <sip:6046370912@office2.voipportal.ca>

Call-ID: go2zi5t9@pbx

CSeq: 31263 NOTIFY

Max-Forwards: 70

Contact: <sip:208.68.18.228:5060;transport=udp>

Event: message-summary

Subscription-State: terminated;reason=noresource

Content-Type: application/simple-message-summary

Content-Length: 103

 

Messages-Waiting: no

Message-Account: sip:6046370912@office2.voipportal.ca

Voice-Message: 0/0 (0/0)

 

 

[9] 2008/04/25 07:54:01: SIP Tr udp:207.6.229.160:60778:

CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport

From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482

To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca>

Call-ID: 5dc9a996@pbx

CSeq: 27432 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[9] 2008/04/25 07:54:03: Last message repeated 2 times

 

 

[9] 2008/04/25 07:54:03: Message repetition, packet dropped

[9] 2008/04/25 07:54:07: SIP Tr udp:207.6.229.160:60778:

NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0

Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport

From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c

To: Netfone Telesales <sip:6046370912@office2.voipportal.ca>

Call-ID: go2zi5t9@pbx

CSeq: 31263 NOTIFY

Max-Forwards: 70

Contact: <sip:208.68.18.228:5060;transport=udp>

Event: message-summary

Subscription-State: terminated;reason=noresource

Content-Type: application/simple-message-summary

Content-Length: 103

 

Messages-Waiting: no

Message-Account: sip:6046370912@office2.voipportal.ca

Voice-Message: 0/0 (0/0)

Link to post
Share on other sites
RTP-RxStat: Dur=25,Pkt=259,Oct=8288,Underun=0

RTP-TxStat: Dur=5,Pkt=259,Oct=8288

 

That indicates that the PBX received the same amount as it send (probably the 20 seconds time difference come from ringing). From that persoective, it does not "sound" like one-way audio problem.

 

The packet size for the 200 okay is in the 1200 bytes range. There are only 300 bytes left until you will experience UDP fragmentation. That might be a problem when you offer more codecs or add another Record-Route into the packet (or just dial a very long name).

 

Also check http://wiki.pbxnsip.com/index.php/One-way_Audio.

 

It would probably better to split this topic off to a new topic, as it is not directly related to the 2.1.7 release.

Link to post
Share on other sites

Hi,

 

I think that there is a mis-understanding of the problem.

 

The problem is that the ATA does not see the call at all. The phone does not ring at the ATA. It is not an issue of one way audio. This only happens on boxes where we migrated from 1.5 to 2.0. If we registers the device on a clean 2.0 PBXnSIP, it works as a charm.

 

Something that may be related to this is that when we restart the PBXnSIP, we lose all the device registrations. Meanwhile if we do the same on a clean 2.0 install, this does not happen.

 

This is causing us major headaches!!!

 

This is not an issue with one ATA. We have had to move dozens of accounts to a PBX with a new 2.0 install but this is causing major headaches!

 

Rafeh Hulays

 

 

 

That indicates that the PBX received the same amount as it send (probably the 20 seconds time difference come from ringing). From that persoective, it does not "sound" like one-way audio problem.

 

The packet size for the 200 okay is in the 1200 bytes range. There are only 300 bytes left until you will experience UDP fragmentation. That might be a problem when you offer more codecs or add another Record-Route into the packet (or just dial a very long name).

 

Also check http://wiki.pbxnsip.com/index.php/One-way_Audio.

 

It would probably better to split this topic off to a new topic, as it is not directly related to the 2.1.7 release.

Link to post
Share on other sites
2.1.8 is available. The only thing changed was a neccessary fix in the camp on feature. Workaround is to disable camp on or upgrade to 2.1.8. See the release notes at http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.8.

 

 

I'm using 2.1.6.2450 - and I could not duplicate the issue fixed in 2.1.8.

 

I also tried 2.1.7 - but still could not duplicate.

 

i'm trying to determine if I need to upgrade from 2.1.6 to 2.1.8 to avoid this situation.

 

Please advise.

Link to post
Share on other sites
Why not for "no answer"? I think it is nice if someone is out for lunch, you want to talk to him and get a call back after he is back from lunch and finishes the first call.

 

The problem is they might not come back for hours, or might not make a call for hours/days, and by the time the callback happens, the person forgets that they ever requested the camp in the first place.

Link to post
Share on other sites
The problem is they might not come back for hours, or might not make a call for hours/days, and by the time the callback happens, the person forgets that they ever requested the camp in the first place.

 

Okay, that makes sense, but then maybe the solution would be to have some kind of timeout with the callback (e.g. one hour).

Link to post
Share on other sites
Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

Loading...
×
×
  • Create New...