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MOH with Aastra 9133i not working


Detlef
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Since I upgraded to PBX 2.1.0.xxxx my MOH is not working no more. Now after some testing I think it is the Aastra phone causing this.

 

If I use an extension with a softphone X-Lite and call another extension with an Aastra 9133i phone and put the call on hold from the X-Lite softphone then the Aastra plays music on hold (from default file). Do I do it the oposite way and put the call on hold from the Aastra 9133i phone then its silent and no MOH plays.

 

The same happens when I call in over the PSTN gateway. If I call the extension with the X-Lite softphone and put that call on hold from the softphone then the caller hears music on hold. If I call the extension with the Aastra phone and put it on hold then its silent to the caller over the PSTN line.

 

Is that a setting in the Aastra phones that I need to correct? It was working with PBXnSIP 2.0.3.1715 and quit with the new 2.1.0.xxxx versions. Since the X-Lite softphone does it correctly I think it has to do with the Aastra phones themself.

 

Currently I am running PBXnSIP 2.1.0.2114 and the Aastra 9133i is firmware 1.4.2.1081

 

Detlef

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Works here, using PBX 2.1.0.2115 and Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5. Maybe you need to upgrade the PBX to the 2115 build.

 

Will do tonight, but since I upgraded to any previous 2.1.0.xxxx version the 9133i phones here quit doing MOH. Going back to 2.0.3.1715 and the MOH always started working again. What I dont understand is that the X-Lite softphone does it correct. I went to the newst Aastra firmware available for download but no change - still no MOH.

 

Below my MAC.CFG everything else on the Aastra settings is left to default:

 

download protocol: TFTP

tftp server: pbx.ims-va.com

auto resync mode: 3

auto resync time: 03:00

 

time server disabled: 0

time server1: dc.ims-va.com

 

sip digit timeout: 3

sip dial plan: "x+#|xx+*"

 

sip mode: 0

sip proxy ip: pbx.ims-va.com

sip proxy port: 5060

sip registrar ip: pbx.ims-va.com

sip registrar port: 0

sip registration period: 3600

sip screen name: User Name

sip user name: 101

sip display name: User Name

sip auth name: 101

sip password: xxxxxx

 

sip vmail: *97

 

directory 1: companylist.csv

directory 2: externallist.csv

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Works here, using PBX 2.1.0.2115 and Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5. Maybe you need to upgrade the PBX to the 2115 build.

 

Ok done, upgraded to .2115 and the 9133i Aastra phones still do not initiate MOH when putting someone on hold or parking a call.

 

Is someone using 9133i phones who has it working? Maybe we could exchange config files??? Would be really appreciated!!

 

Detlef

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Do you have the SIP INVITE that tells the PBX to hold the call? Maybe the phone is using the old style with 0.0.0.0 (which was obsoleted in June 2002, see RFC 3264)?

 

This is the logfile I see as soon as I press the HOLD button on the Aastra phone during a connected call with the X-Lite softphone:

 

192.168.104.222 = Aastra Phone

192.168.104.129 = X-Lite Softphone

192.168.104.220 = PBXnSIP

 

 

[9] 2007/10/10 08:55:20: SIP Rx udp:192.168.104.222:5060:

INVITE sip:101@192.168.104.220:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bKdd007bf64

Max-Forwards: 70

Content-Length: 266

To: "X-Lite" <sip:150@localhost>;tag=43086

From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a

Call-ID: e4afabe4@pbx

CSeq: 69380235 INVITE

Supported: timer

Allow-Events: talk,hold,conference

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

Content-Type: application/sdp

Contact: Aastra <sip:101@192.168.104.222>

Supported: replaces

User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

 

v=0

o=MxSIP 0 1152796867 IN IP4 192.168.104.222

s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 3000 RTP/AVP 0 8 18 2 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[9] 2007/10/10 08:55:20: Resolve destination 2684: a udp 192.168.104.222 5060

[9] 2007/10/10 08:55:20: Resolve destination 2684: udp 192.168.104.222 5060

[9] 2007/10/10 08:55:20: SIP Tx udp:192.168.104.222:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bKdd007bf64

From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a

To: "X-Lite" <sip:150@localhost>;tag=43086

Call-ID: e4afabe4@pbx

CSeq: 69380235 INVITE

Contact: <sip:101@192.168.104.220:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.0.2115

Content-Type: application/sdp

Content-Length: 275

 

v=0

o=- 34981 34981 IN IP4 192.168.104.220

s=-

c=IN IP4 192.168.104.220

t=0 0

m=audio 64154 RTP/AVP 0 8 18 2 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:18 g729/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

 

[9] 2007/10/10 08:55:20: SIP Rx udp:192.168.104.222:5060:

ACK sip:101@192.168.104.220:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK31b41b11a

Max-Forwards: 70

Content-Length: 0

To: "X-Lite" <sip:150@localhost>;tag=43086

From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a

Call-ID: e4afabe4@pbx

CSeq: 69380235 ACK

Contact: Aastra <sip:101@192.168.104.222>

User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

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Oh yea, they are using the 0.0.0.0. I am not sure if we should still support this more than 5 years old "workaround" - isn't there a SW upgrade for the phone available?

 

I just put the latest and greatest firmware on the phone that was available from their website...

 

AASTRA TELECOM INC.

June 2007

Generic SIP Firmware 1.4.2.1081 GA Release.

FC-000032-01-11 480i

FC-000040-00-11 480iCT

FC-000046-01-11 9133i

FC-000058-01-11 9112i

 

Maybe I can have them change it if I contact their customer support and ask for the correct ON HOLD procedure? What would I need to tell them to see if they would change it to the up-to-date way?

 

EDIT:

I just sent them an email... lets see what they will say

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Nice, I got a reply from Aastra. They have this problem on their feature request list... hopefully not since 5 years!!

 

From: Layne Monson [mailto:layne.monson@aastra.com]Sent: Wednesday, October 10, 2007 5:33 PM

To: Detlef Schade

Subject: IP11755: Aastra 9133i - ON HOLD Problem with 0.0.0.0 INVITE

 

Detlef,

This had already been submitted as a feature request for these phones . Other methods are not supported right now.

 

Thanks,

 

Layne Monson CCNA

Customer Support Engineer II

Aastra Intecom

2811 Internet Blvd.

Frisco, TX 75034

 

layne.monson@aastra.com

469 365 3847 direct

 

EDIT:

 

Oh, just got an accurate estimate - its even worse than 5 years, they have no clue when they will update their phones:

 

There is no way I can say how long it would be. It?s totally up to upper management to decide what features and fixes go into each load and allocate dev resources.

 

Sorry , there is nothing I can do further.

 

Thanks,

 

Layne Monson CCNA

Customer Support Engineer II

Aastra Intecom

2811 Internet Blvd.

Frisco, TX 75034

 

layne.monson@aastra.com

469 365 3847 direct

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Hmm. Maybe it is easier to program around it and support that old style as well.

 

I would really appreciate that, currently I am stuck with the new PBX not supporting it any more and Aastra not knowing if or when they update their phones... the pain ist that it is silent to the caller if you put someone on hold with the Aastra phone and they think the line dropped and mostly hang up.

 

I guess their latest firmware is used in all 4 of those phones listed in the release info?

 

FC-000032-01-11 480i

FC-000040-00-11 480iCT

FC-000046-01-11 9133i

FC-000058-01-11 9112i

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Do you already know if you will have a PBX version 2.1.0.2116 that will support this historic on hold procedures again??

 

Okay there is a 2116 version (http://www.pbxnsip.com/download/pbxctrl-2.1.0.2116.exe), and we did a small change in the 0.0.0.0 detection, but we had no chance of trying it out. Please give it a try.

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Put the .2116 in this morning but there seems to be a bug - its still not working! It keeps asking for "Authentication Required" and reports "Password does not match". Below the logfile with a call from a X-Lite softphone to an Aastra 9133i when the Aastra was trying to put the X-Lite on hold:

 

 

 

[9] 2007/10/15 07:43:51: SIP Rx udp:192.168.104.222:5060:

INVITE sip:101@192.168.104.220:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK24eefb4a0

Max-Forwards: 70

Content-Length: 265

To: "X-Lite Softphone" <sip:150@localhost>;tag=27498

From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d

Call-ID: 1e63199b@pbx

CSeq: 953138503 INVITE

Supported: timer

Allow-Events: talk,hold,conference

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

Content-Type: application/sdp

Contact: D.Schade <sip:101@192.168.104.222>

Supported: replaces

User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

 

v=0

o=MxSIP 0 369883472 IN IP4 192.168.104.222

s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 3000 RTP/AVP 0 8 18 2 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[9] 2007/10/15 07:43:51: Resolve destination 190: a udp 192.168.104.222 5060

[9] 2007/10/15 07:43:51: Resolve destination 190: udp 192.168.104.222 5060

[9] 2007/10/15 07:43:51: SIP Tx udp:192.168.104.222:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK24eefb4a0

From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d

To: "X-Lite Softphone" <sip:150@localhost>;tag=27498

Call-ID: 1e63199b@pbx

CSeq: 953138503 INVITE

Contact: <sip:101@192.168.104.220:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.0.2116

Content-Type: application/sdp

Content-Length: 273

 

v=0

o=- 6791 6791 IN IP4 192.168.104.220

s=-

c=IN IP4 192.168.104.220

t=0 0

m=audio 54610 RTP/AVP 0 8 18 2 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:18 g729/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2007/10/15 07:43:52: SIP Rx udp:192.168.104.222:5060:

ACK sip:101@192.168.104.220:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK00cb18415

Max-Forwards: 70

Content-Length: 0

To: "X-Lite Softphone" <sip:150@localhost>;tag=27498

From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d

Call-ID: 1e63199b@pbx

CSeq: 953138503 ACK

Contact: D.Schade <sip:101@192.168.104.222>

User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

 

 

[9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055:

SUBSCRIBE sip:192.168.104.220 SIP/2.0

Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport

From: <sip:101@ims-va.com>;tag=2766

To: <sip:101@ims-va.com>

Call-ID: zycdrj93q2jke09ah8rj

CSeq: 10551 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:101@0.0.0.0:1055>

Event: x-tapi

Accept: application/x-tapi

Expires: 3600

Content-Length: 0

 

 

[9] 2007/10/15 07:43:57: Last message repeated 2 times

[9] 2007/10/15 07:43:57: Message repetition, packet dropped

[9] 2007/10/15 07:43:57: Resolve destination 192: udp 192.168.104.129 1055

[9] 2007/10/15 07:43:57: SIP Tx udp:192.168.104.129:1055:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129

From: <sip:101@ims-va.com>;tag=2766

To: <sip:101@ims-va.com>;tag=0b4a71dab2

Call-ID: zycdrj93q2jke09ah8rj

CSeq: 10551 SUBSCRIBE

User-Agent: pbxnsip-PBX/2.1.0.2116

WWW-Authenticate: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",domain="sip:192.168.104.220",algorithm=MD5

Content-Length: 0

 

 

[9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055:

SUBSCRIBE sip:192.168.104.220 SIP/2.0

Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport

From: <sip:101@ims-va.com>;tag=2766

To: <sip:101@ims-va.com>

Call-ID: zycdrj93q2jke09ah8rj

CSeq: 10552 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:101@0.0.0.0:1055>

Event: x-tapi

Accept: application/x-tapi

Authorization: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",response="067ab894d4b5db7e1cc860d426c2109d",username="101",uri="sip:192.168.104.220",algorithm=MD5

Expires: 3600

Content-Length: 0

 

 

[9] 2007/10/15 07:43:57: Resolve destination 193: udp 192.168.104.129 1055

[9] 2007/10/15 07:43:57: SIP Tx udp:192.168.104.129:1055:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129

From: <sip:101@ims-va.com>;tag=2766

To: <sip:101@ims-va.com>;tag=0b4a71dab2

Call-ID: zycdrj93q2jke09ah8rj

CSeq: 10552 SUBSCRIBE

User-Agent: pbxnsip-PBX/2.1.0.2116

Warning: 399 ims-va.com Password does not match

Content-Length: 0

 

 

[9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055:

SUBSCRIBE sip:192.168.104.220 SIP/2.0

Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport

From: <sip:101@ims-va.com>;tag=2766

To: <sip:101@ims-va.com>

Call-ID: zycdrj93q2jke09ah8rj

CSeq: 10552 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:101@0.0.0.0:1055>

Event: x-tapi

Accept: application/x-tapi

Authorization: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",response="067ab894d4b5db7e1cc860d426c2109d",username="101",uri="sip:192.168.104.220",algorithm=MD5

Expires: 3600

Content-Length: 0

 

 

[9] 2007/10/15 07:43:57: SIP Tm udp:192.168.104.129:1055:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129

From: <sip:101@ims-va.com>;tag=2766

To: <sip:101@ims-va.com>;tag=0b4a71dab2

Call-ID: zycdrj93q2jke09ah8rj

CSeq: 10552 SUBSCRIBE

User-Agent: pbxnsip-PBX/2.1.0.2116

Warning: 399 ims-va.com Password does not match

Content-Length: 0

 

 

[9] 2007/10/15 07:43:57: Message repetition, packet dropped

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Okay, we did a simulation without the a= header on our phone here and maybe the following version fixes that problem:

 

http://www.pbxnsip.com/download/pbxctrl-2.1.0.2117.exe

 

We found something that would explain why the 0.0.0.0 method would not work. Please verify.

 

This version did the trick!! Yeah... my MoH is working again with the 9133i phones!!

 

Thanks alot for the fast help adapting to the old style!

 

Detlef

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