YMSL
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Posts posted by YMSL
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There is a SOAP request for this (especially for hospitality purposes). Not sure if that is "simple"...
Can you tell me where to look for?
Is their a list of code for that somewhere?
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If I want to use a particular extension like a guest room, is their any simple process to reset completelly the mailbox without deleting the extension and recreate it? Something like an hospitality system!
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Many thanks for your reply, I will order 1 unit and test it. I'll keep the forum posted
Test done,
We have received from Cisco SPA525G unit and put them under testing.
result:
GRATE PHONE, look, size, feel... very handsome device
handfree seeker is so fantastic
so fare interoperability seams to be just fine
provisionning need be be done partially manually...
needs lots of manual adjustment for Wi-Fi usability
USB (mp3) player need standard mp3 files
need firmware upgrade from 7.1.3 to 7.1.7 (A MOST)
-UPDATE- Feb 2nd
SPA525G product manager from California is providing support,
so it look like cisco is getting very very serious with this promessing device.
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If you go into the Registrar page you can add a contact number at the bottom of the page.
information would be something like
sip:5557271111@IPAddressOfRegistrar
This then makes the system think the cell phone is a registered extension and calls it at the same time it calls the IP Phone. Plus it works in a Hunt group.
Each outgoing call to a cell phone uses a trunk, once one cell phone or IP phone is connected the other trunks are released.
This feature works on version 2 of pbxnsip. Not so sure about 3+.
For me in 3.2 is does not...
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Dear YMSL,
The Mediatrix 1204 puts, by default, the CID detected on the analog side. If it doesn't detect ANY CallerID, it will default to the configured FriendlyName and Username.
There are three options :
You have configure the 1204 to seize the line before CallerID is detected (second ring), there are no CallerID on this line, the 1204 cannot detect the CallerID.
I know of one case where the 1204 was plugged into a very old PSTN switch that had a old CallerID standard. But it is very rare.
To check if the 1204 is configured to seize the line before the CallerID is detected, look at the fxoIfAnsweringDelayTable.
1. Make sure the fxoAnswerOnCallerIdDetectionEnable is Enable
2. Make sure the fxoPreAnswerDelay is high enough (8000 should be more then enough)
And last, try a regular phone to see if you get CallerID
Good luck !
thanks for that information it is very valuable.
All lines that are in used were on a PSTN system before and didi displayed CID for all calls.
fxoAnswerOnCallerIdDetectionEnable is Enable
fxoPreAnswerDelay is at 8000
and still not getting anything.
any other idea?
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Hi YMSL,
Try using outOfBandSignalingProtocol
but also set the following :
sipInteropDtmfTransportMethod to "1: infoDtmfRelay"
Mediatrix units use an old draft of SIP INFO by default. (aka Choudhuri)
Many thanks,
this worked - I am gratefull
Now, Caller ID is not passed to PBXnSIP , any idea why? All CID are showed like they were comming from the mediatrix registration (USER and Frendlly name) so my number!
many thanks
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PLEASE - URGENT
I have major problem with that gateway, and I really need help.
Mediatrix 1204 sip V5
DTMF is not working - income calls cannot dial extensions or anything else on AT or extensions, I have tried
InBand, outOfBandUsignRTP, outOfBandSignalingProtocol, SignalingProtocolDependent.
AllowDTMF is active.
Where should I be looking for?
What I do not get is that LinkSys SPA3102 is working just fine and Mediatrix that is "better" product is just runing me crazy.
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It might no be a question at all...
Does Mac Installer for Darwin 9.0 will run under Darwin 9.6 If not do you have plans to release it?
regards
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Hmm. A search for "LDAP" does not return a result...
Look at this one P 71-74 This is an extract of the manual
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Oh I did not know they support LDAP... That would be another way of reason the address book! Any links to that?
the only information I can find is from LinkSys web Site at:
http://www-ca.linksys.com/servlet/Satellit...d=1519652913B14
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Well, I don't see a lot of traffic, but no NOTIFY... Either that is the problem or the logging wasn't turned on the that. It would be good to see also the SUBSCRIBE for the resource.
I have Logs settings all "on" (everything)
Please see the attached file for logs details
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That is a difficult topic.
First, the question is: Who wants to read the directory?
If it is the phone, then we have a couple of methods. Polycom gets the address book provisioned after the reboot, snom pulls it down in its XML format (via HTTP), and there are new ideas about using XCAP for a more interoperable way of having an address book.
If it is the PBX that should pull down the directory (especially from a ActiveDirectory), that is a completely different topic.
And then we have a new topic coming up. LinkedIn, Xing, Salesforce, and I-don't-know what social networks are storing a lot of very interesting data (even with pictures!) that users would love to see on the display when a call comes in. A standard? No way. Everybody doing it in a different way. That is not making our life easier...
Many thanks for your reply, I understand, it is a large topic. So to make it simple let me rephrase what I need.
All phones are LinkSys SPA962/SPA942
PBX is running under WinBox
Idealy, I would like PBX to generate whatever it need in order to have all SPA phones to reade and display Company directory.
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Running 3.1.2 on Windows box
1- Does PBXnSIP support Company Directory in some ways? and does it generate content that can be acessible company wide? If yes How?
If not
2- Does LDAP could be a good solution? and How
My goal is to get everyone access to the complete company directory on every phone, either by grabbing some file from PBX or true provisioning (ould be better)
tkx
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You need to log the SIP packets, SUBSCRIBE/NOTIFY. They are usually logged on level 7, unless you monitor a specifc IP address.
here it is:
Log In
[7] 2009/01/14 13:20:27: SIP Rx udp:10.0.1.1:44143:
INVITE sip:*64@10.0.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-d5380752
From: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
To: "Agent Log" <sip:*64@10.0.1.1>
Call-ID: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Test User1" <sip:22@10.0.1.5:5060>
Expires: 240
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 385
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 46919 46919 IN IP4 10.0.1.5
s=-
c=IN IP4 10.0.1.5
t=0 0
m=audio 61042 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[7] 2009/01/14 13:20:27: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 100 Trying
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-d5380752;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
i: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 101 INVITE
l: 0
[7] 2009/01/14 13:20:27: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 401 Authentication Required
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-d5380752;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
i: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 101 INVITE
User-Agent: pbxnsip-PBX/3.1.2.3120
WWW-Authenticate: Digest realm="10.0.1.1",nonce="8459ee06553585fc7eef947239f58021",domain="sip:*64@10.0.1.1",algorithm=MD5
l: 0
[7] 2009/01/14 13:20:27: SIP Rx udp:10.0.1.1:44143:
ACK sip:*64@10.0.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-d5380752
From: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
To: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
Call-ID: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Test User1" <sip:22@10.0.1.5:5060>
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 0
[7] 2009/01/14 13:20:27: SIP Rx udp:10.0.1.1:44143:
INVITE sip:*64@10.0.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-947ebfc2
From: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
To: "Agent Log" <sip:*64@10.0.1.1>
Call-ID: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="22",realm="10.0.1.1",nonce="8459ee06553585fc7eef947239f58021",uri="sip:*64@10.0.1.1",algorithm=MD5,response="193078f40553d959c5ff7278c0e479c7"
Contact: "Test User1" <sip:22@10.0.1.5:5060>
Expires: 240
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 385
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 46919 46919 IN IP4 10.0.1.5
s=-
c=IN IP4 10.0.1.5
t=0 0
m=audio 61042 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[7] 2009/01/14 13:20:27: Set packet length to 20
[6] 2009/01/14 13:20:27: Sending RTP for bd49bbbc-d8d3013a@10.0.1.5#45e65484b2 to 10.0.1.5:61042
[7] 2009/01/14 13:20:27: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 100 Trying
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-947ebfc2;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
i: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 102 INVITE
l: 0
[5] 2009/01/14 13:20:27: Agent 22@192.168.0.1 logged in
[7] 2009/01/14 13:20:27: Set packet length to 20
[7] 2009/01/14 13:20:27: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 200 Ok
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-947ebfc2;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
i: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 102 INVITE
m: <sip:22@10.0.1.1:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.1.2.3120
c: application/sdp
l: 300
v=0
o=- 992 992 IN IP4 10.0.1.1
s=-
c=IN IP4 10.0.1.1
t=0 0
m=audio 57988 RTP/AVP 0 8 18 2 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[7] 2009/01/14 13:20:27: Last message repeated 2 times
[7] 2009/01/14 13:20:27: SIP Rx udp:10.0.1.1:44143:
ACK sip:22@10.0.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-27875cf1
From: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
To: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
Call-ID: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="22",realm="10.0.1.1",nonce="8459ee06553585fc7eef947239f58021",uri="sip:*64@10.0.1.1",algorithm=MD5,response="193078f40553d959c5ff7278c0e479c7"
Contact: "Test User1" <sip:22@10.0.1.5:5060>
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 0
[7] 2009/01/14 13:20:27: Last message repeated 2 times
[6] 2009/01/14 13:20:27: Sending RTP for bd49bbbc-d8d3013a@10.0.1.5#45e65484b2 to 10.0.1.1:33302
[7] 2009/01/14 13:20:27: Receiving DTMF on codec 101
[7] 2009/01/14 13:20:29: SIP Tx udp:10.0.1.1:44143:
BYE sip:22@10.0.1.5:5060 SIP/2.0
v: SIP/2.0/UDP 10.0.1.1:5060;branch=z9hG4bK-3b03a83bf01942c3e4d9057aa69b0539;rport
f: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
t: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
i: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 1392 BYE
Max-Forwards: 70
m: <sip:22@10.0.1.1:5060>
RTP-RxStat: Dur=1,Pkt=68,Oct=11696,Underun=0
RTP-TxStat: Dur=1,Pkt=65,Oct=11180
l: 0
[7] 2009/01/14 13:20:29: SIP Rx udp:10.0.1.1:44143:
SIP/2.0 200 OK
t: "Test User1" <sip:22@10.0.1.1>;tag=d94749d42b590a02o0
f: "Agent Log" <sip:*64@10.0.1.1>;tag=45e65484b2
i: bd49bbbc-d8d3013a@10.0.1.5
CSeq: 1392 BYE
v: SIP/2.0/UDP 10.0.1.1:5060;branch=z9hG4bK-3b03a83bf01942c3e4d9057aa69b0539
Server: Linksys/SPA962-6.1.3(a)
Content-Length: 0
[7] 2009/01/14 13:20:29: Call bd49bbbc-d8d3013a@10.0.1.5#45e65484b2: Clear last request
[5] 2009/01/14 13:20:29: BYE Response: Terminate bd49bbbc-d8d3013a@10.0.1.5
Log Out
[7] 2009/01/14 13:19:26: SIP Rx udp:10.0.1.1:44143:
INVITE sip:*64@10.0.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-a0743bf1
From: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
To: "Agent Log" <sip:*64@10.0.1.1>
Call-ID: db814912-b65dcc15@10.0.1.5
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Test User1" <sip:22@10.0.1.5:5060>
Expires: 240
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 385
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 40832 40832 IN IP4 10.0.1.5
s=-
c=IN IP4 10.0.1.5
t=0 0
m=audio 61040 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[7] 2009/01/14 13:19:26: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 100 Trying
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-a0743bf1;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
i: db814912-b65dcc15@10.0.1.5
CSeq: 101 INVITE
l: 0
[7] 2009/01/14 13:19:26: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 401 Authentication Required
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-a0743bf1;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
i: db814912-b65dcc15@10.0.1.5
CSeq: 101 INVITE
User-Agent: pbxnsip-PBX/3.1.2.3120
WWW-Authenticate: Digest realm="10.0.1.1",nonce="27dcfb81c5b5fac696203b03e441ac0d",domain="sip:*64@10.0.1.1",algorithm=MD5
l: 0
[7] 2009/01/14 13:19:26: SIP Rx udp:10.0.1.1:44143:
ACK sip:*64@10.0.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-a0743bf1
From: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
To: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
Call-ID: db814912-b65dcc15@10.0.1.5
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Test User1" <sip:22@10.0.1.5:5060>
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 0
[7] 2009/01/14 13:19:26: SIP Rx udp:10.0.1.1:44143:
INVITE sip:*64@10.0.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-383ae20d
From: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
To: "Agent Log" <sip:*64@10.0.1.1>
Call-ID: db814912-b65dcc15@10.0.1.5
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="22",realm="10.0.1.1",nonce="27dcfb81c5b5fac696203b03e441ac0d",uri="sip:*64@10.0.1.1",algorithm=MD5,response="4e5fdc5b0226f2c4977b5d838242a66a"
Contact: "Test User1" <sip:22@10.0.1.5:5060>
Expires: 240
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 385
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 40832 40832 IN IP4 10.0.1.5
s=-
c=IN IP4 10.0.1.5
t=0 0
m=audio 61040 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[7] 2009/01/14 13:19:26: Set packet length to 20
[6] 2009/01/14 13:19:26: Sending RTP for db814912-b65dcc15@10.0.1.5#e480ae1024 to 10.0.1.5:61040
[7] 2009/01/14 13:19:26: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 100 Trying
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-383ae20d;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
i: db814912-b65dcc15@10.0.1.5
CSeq: 102 INVITE
l: 0
[5] 2009/01/14 13:19:26: Agent 22@192.168.0.1 logged out
[7] 2009/01/14 13:19:26: Set packet length to 20
[7] 2009/01/14 13:19:26: SIP Tx udp:10.0.1.1:44143:
SIP/2.0 200 Ok
v: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-383ae20d;rport=44143;received=10.0.1.1
f: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
t: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
i: db814912-b65dcc15@10.0.1.5
CSeq: 102 INVITE
m: <sip:22@10.0.1.1:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.1.2.3120
c: application/sdp
l: 304
v=0
o=- 22278 22278 IN IP4 10.0.1.1
s=-
c=IN IP4 10.0.1.1
t=0 0
m=audio 56528 RTP/AVP 0 8 18 2 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[7] 2009/01/14 13:19:26: Last message repeated 2 times
[7] 2009/01/14 13:19:26: SIP Rx udp:10.0.1.1:44143:
ACK sip:22@10.0.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-d46f0f5b
From: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
To: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
Call-ID: db814912-b65dcc15@10.0.1.5
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="22",realm="10.0.1.1",nonce="27dcfb81c5b5fac696203b03e441ac0d",uri="sip:*64@10.0.1.1",algorithm=MD5,response="4e5fdc5b0226f2c4977b5d838242a66a"
Contact: "Test User1" <sip:22@10.0.1.5:5060>
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 0
[6] 2009/01/14 13:19:26: Sending RTP for db814912-b65dcc15@10.0.1.5#e480ae1024 to 10.0.1.1:44917
[7] 2009/01/14 13:19:26: Receiving DTMF on codec 101
[7] 2009/01/14 13:19:26: SIP Rx udp:10.0.1.1:44143:
ACK sip:22@10.0.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.5:5060;branch=z9hG4bK-d46f0f5b
From: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
To: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
Call-ID: db814912-b65dcc15@10.0.1.5
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="22",realm="10.0.1.1",nonce="27dcfb81c5b5fac696203b03e441ac0d",uri="sip:*64@10.0.1.1",algorithm=MD5,response="4e5fdc5b0226f2c4977b5d838242a66a"
Contact: "Test User1" <sip:22@10.0.1.5:5060>
User-Agent: Linksys/SPA962-6.1.3(a)
Content-Length: 0
[7] 2009/01/14 13:19:28: SIP Tx udp:10.0.1.1:44143:
BYE sip:22@10.0.1.5:5060 SIP/2.0
v: SIP/2.0/UDP 10.0.1.1:5060;branch=z9hG4bK-e7bfb34b595b3c02257b7de0abe51546;rport
f: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
t: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
i: db814912-b65dcc15@10.0.1.5
CSeq: 26363 BYE
Max-Forwards: 70
m: <sip:22@10.0.1.1:5060>
RTP-RxStat: Dur=1,Pkt=63,Oct=10836,Underun=0
RTP-TxStat: Dur=1,Pkt=64,Oct=11008
l: 0
[7] 2009/01/14 13:19:28: SIP Rx udp:10.0.1.1:44143:
SIP/2.0 200 OK
t: "Test User1" <sip:22@10.0.1.1>;tag=92f5eceeaeb019e9o0
f: "Agent Log" <sip:*64@10.0.1.1>;tag=e480ae1024
i: db814912-b65dcc15@10.0.1.5
CSeq: 26363 BYE
v: SIP/2.0/UDP 10.0.1.1:5060;branch=z9hG4bK-e7bfb34b595b3c02257b7de0abe51546
Server: Linksys/SPA962-6.1.3(a)
Content-Length: 0
[7] 2009/01/14 13:19:28: Call db814912-b65dcc15@10.0.1.5#e480ae1024: Clear last request
[5] 2009/01/14 13:19:28: BYE Response: Terminate db814912-b65dcc15@10.0.1.5
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Hmm. Maybe you can post the XML documents that the PBX sends to the phone when the state changes (the body of the NOFIFY messages). Then we can see if the phone should believe that the resource is "active".
Not a problem, what level of logs will do? certenally not "0"
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Ehhhh.... Okay, maybe temporarily turn Javascript off. Or you can also use a handcraftet URL like this: http://yourpbx/dom_feature_codes.htm?editd...gent_logout=*75.
This does the trick. Only one little down, Lamp is not changing color either if logedin or out
any idea on where to look?
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The code for the login and logout may be the same. If that is the case, then calling that code will toggle the state. Maybe that helps?
This could have been the solution BUT, Web interface does not allow to have duplication code, If I try the web interface is telling me "Dupilcate"
Can we change the feature code someplace else?
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Runnig 3.1.2 on Windows Box - CallCenter licence
Now that agent can log in and out I am trying to use a Key on our SAP962/SPA942 Agent phone to display status, either loged in or loged out just like a service flag will do.
for service flag we are usign
"fnc=sd+blf+cp;nme=Reception;sub=30@192.168.0.1;usr=30@192.168.0.1;vid=1"
if service flag is "clear" Lamp is Green
if service flag is "Set" lamp is Red
I tried the same wuth agent presence but since the it is usign "*64/*65" I need 2 phone buttons and it is not working well.
Any idea on how to create a beaviour or a function that will accomplish the same visual result
If agent is logged "in" lamp will go Green
If Agent is logges "out" lamp will go Red
Looking at the XML file I can view agent status, but is is not very elegent!
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We are setting our system in a hold fashen manner, ...
Usign a Service Flag account to have a humain person getting incoming call during business houres and if outside business houres we want it to go to a different AT.
and it does not work.
settings:
AT 100 - open houres set with Service Flag Account 150 - Night Service Number 101
AT 101 - close houres
EX front desk 110
SF 150 - set with business hours and holidays
what am I missing?
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That is a kind of Linksys classical problem. The default dial plan of the phone makes it hard to dial star codes on the PBX. See http://wiki.pbxnsip.com/index.php/Linksys for some dial plan examples.
On my side I use (the system is usgn 4 digets extensions)
(*xxxxxx*x|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
and it work just fine. I've alos discover that if I remove all "*xx" features code on the phone, it use the feature code from PBX and NOT from the phone.
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Well, at least Broadsoft is using SIP, and I guess with Asterisk they are also using SIP so I would say it works. But only the real device will tell how much exactly works.
Many thanks for your reply, I will order 1 unit and test it. I'll keep the forum posted
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Well if they keep their software it should work like to other models...
"The SPA525G ... such as the Cisco SPA9000 Voice System or a Broadsoft or Asterisk system" from their user manual PDF, is their any way we can make that certain?
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Does anybody know if the new Cisco/LinkSys SPA525G is or will be fully operational on PXBnSIP 3.1.1
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Hmm. The PBX generates the files for SPA941. They should be pretty much the same?
It seams to be working but I cannot control all parameters, i.e. multiple lines and other spa962 specific settings.
Also PBX do not write the generated file anyware. TFTP is set to "on" - "To file" - "Always"
3.2.0
in New Features/Versions
Posted
Version 3.2.0 seams to have a little problem, we are no longer capable to use Call Intrusion functions. fo my costumer on a call center version it is very dramatic.
Go to go back to 3.1.2
and everything went back to normal