badgewick
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Posts posted by badgewick
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We currently have a CS410 that is running Linux but I'm unfamiliar with how to access it other than with telnet from a Windows workstation. I've attempted to connection and I cannot get a telnet session up. Any help would be appreciated.
This system was already running with the Linux OS, but it's not in production yet so if someone can instruct me on how to convert I don't have a problem with that either.
Edit: I've also attempted to connect via FTP and SSH and neither are letting me connect to do the software update.
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Check out http://wiki.pbxnsip.com/index.php/Installi...oftware_updates. After updating to 2084, you should update to 2115 (see http://www.pbxnsip.com/software).
I've read that post but I'm not sure how to access the linux os to be able to update it.
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We can provide external access for someone to get in, but I can't find any phone numbers or email addresses in which to open a private trouble ticket.
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Here is the log file I recieved. I don't THINK it's a routing problem. The box was power cycled and now we can't even make out going calls. Everything gets a busy signal. The following log was to an internal extension.
[7] 2007/11/05 16:31:01: SIP Rx udp:192.168.1.103:5060:
INVITE sip:101@192.168.1.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bKbd2515bd33876D6E
From: "Counselor2" <sip:103@192.168.1.10>;tag=293C3E8F-53FF28DA
To: <sip:101@192.168.1.10;user=phone>
CSeq: 1 INVITE
Call-ID: 432735b3-b2286179-9e07911c@192.168.1.103
Contact: <sip:103@192.168.1.103>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/2.1.2.0078
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 227
v=0
o=- 1194295822 1194295822 IN IP4 192.168.1.103
s=Polycom IP Phone
c=IN IP4 192.168.1.103
t=0 0
m=audio 2228 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
[7] 2007/11/05 16:31:01: UDP: Opening socket on port 51638
[7] 2007/11/05 16:31:01: UDP: Opening socket on port 51639
[8] 2007/11/05 16:31:01: Could not find a trunk (1 trunks)
[9] 2007/11/05 16:31:01: Resolve destination 222: a udp 192.168.1.103 5060
[9] 2007/11/05 16:31:01: Resolve destination 222: udp 192.168.1.103 5060
[7] 2007/11/05 16:31:01: SIP Tx udp:192.168.1.103:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bKbd2515bd33876D6E
From: "Counselor2" <sip:103@192.168.1.10>;tag=293C3E8F-53FF28DA
To: <sip:101@192.168.1.10;user=phone>;tag=6b49d561d6
Call-ID: 432735b3-b2286179-9e07911c@192.168.1.103
CSeq: 1 INVITE
Content-Length: 0
[9] 2007/11/05 16:31:04: Resolve destination 223: a udp 192.168.1.103 5060
[9] 2007/11/05 16:31:04: Resolve destination 223: udp 192.168.1.103 5060
[7] 2007/11/05 16:31:04: SIP Tx udp:192.168.1.103:5060:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bKbd2515bd33876D6E
From: "Counselor2" <sip:103@192.168.1.10>;tag=293C3E8F-53FF28DA
To: <sip:101@192.168.1.10;user=phone>;tag=6b49d561d6
Call-ID: 432735b3-b2286179-9e07911c@192.168.1.103
CSeq: 1 INVITE
User-Agent: pbxnsip-PBX/2.0.9.2059
WWW-Authenticate: Digest realm="192.168.1.10",nonce="fdff8a6bdd167e90475a12688a06d89d",domain="sip:101@192.168.1.10:5060;user=phone",algorithm=MD5
Content-Length: 0
[7] 2007/11/05 16:31:04: SIP Rx udp:192.168.1.103:5060:
ACK sip:101@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bKbd2515bd33876D6E
From: "Counselor2" <sip:103@192.168.1.10>;tag=293C3E8F-53FF28DA
To: <sip:101@192.168.1.10;user=phone>;tag=6b49d561d6
CSeq: 1 ACK
Call-ID: 432735b3-b2286179-9e07911c@192.168.1.103
Contact: <sip:103@192.168.1.103>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/2.1.2.0078
Max-Forwards: 70
Content-Length: 0
[7] 2007/11/05 16:31:04: SIP Rx udp:192.168.1.103:5060:
INVITE sip:101@192.168.1.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK1f0e2675FDE44686
From: "Counselor2" <sip:103@192.168.1.10>;tag=293C3E8F-53FF28DA
To: <sip:101@192.168.1.10;user=phone>
CSeq: 2 INVITE
Call-ID: 432735b3-b2286179-9e07911c@192.168.1.103
Contact: <sip:103@192.168.1.103>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/2.1.2.0078
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Authorization: Digest username="103", realm="192.168.1.10", nonce="fdff8a6bdd167e90475a12688a06d89d", uri="sip:101@192.168.1.10:5060;user=phone", response="5562519ae0c8aaf11b75d8d3a43eb86e", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 227
v=0
o=- 1194295822 1194295822 IN IP4 192.168.1.103
s=Polycom IP Phone
c=IN IP4 192.168.1.103
t=0 0
m=audio 2228 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
[8] 2007/11/05 16:31:04: Tagging request with existing tag
[9] 2007/11/05 16:31:04: Resolve destination 224: a udp 192.168.1.103 5060
[9] 2007/11/05 16:31:04: Resolve destination 224: udp 192.168.1.103 5060
[7] 2007/11/05 16:31:04: SIP Tx udp:192.168.1.103:5060:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK1f0e2675FDE44686
From: "Counselor2" <sip:103@192.168.1.10>;tag=293C3E8F-53FF28DA
To: <sip:101@192.168.1.10;user=phone>;tag=6b49d561d6
Call-ID: 432735b3-b2286179-9e07911c@192.168.1.103
CSeq: 2 INVITE
User-Agent: pbxnsip-PBX/2.0.9.2059
Warning: 399 192.168.1.10 Password does not match
Content-Length: 0
[7] 2007/11/05 16:31:04: SIP Rx udp:192.168.1.103:5060:
ACK sip:101@192.168.1.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK1f0e2675FDE44686
From: "Counselor2" <sip:103@192.168.1.10>;tag=293C3E8F-53FF28DA
To: <sip:101@192.168.1.10;user=phone>;tag=6b49d561d6
CSeq: 2 ACK
Call-ID: 432735b3-b2286179-9e07911c@192.168.1.103
Contact: <sip:103@192.168.1.103>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/2.1.2.0078
Max-Forwards: 70
Content-Length: 0
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I just started with this company and working with these systems so please be patient if I don't give all the information you need.
Evidently there was a problem with both of the CS410's they put into service. If they made an outside call it would shut off after a few mins, but internal calls seemed to be ok. When they brought me up to speed on the current situation any call they made took 5 - 10 seconds before it even connected.
Currently I believe I've resolved the problem with the outgoing call by creating a new call plan. However now we can't make internal phone calls with it. Below is the log for the last attempt. Any advice would be appreciated.
[7] 2007/11/05 14:30:09: SIP Rx udp:192.168.1.103:5060:
[7] 2007/11/05 14:30:09: UDP: Opening socket on port 54414
[7] 2007/11/05 14:30:09: UDP: Opening socket on port 54415
[8] 2007/11/05 14:30:09: Could not find a trunk (1 trunks)
[9] 2007/11/05 14:30:09: Resolve destination 6404: a udp 192.168.1.103 5060
[9] 2007/11/05 14:30:09: Resolve destination 6404: udp 192.168.1.103 5060
[7] 2007/11/05 14:30:09: SIP Tx udp:192.168.1.103:5060:
[9] 2007/11/05 14:30:10: Resolve destination 6405: a udp 192.168.1.103 5060
[9] 2007/11/05 14:30:10: Resolve destination 6405: udp 192.168.1.103 5060
[7] 2007/11/05 14:30:10: SIP Tx udp:192.168.1.103:5060:
[7] 2007/11/05 14:30:10: SIP Rx udp:192.168.1.103:5060:
[7] 2007/11/05 14:30:10: SIP Rx udp:192.168.1.103:5060:
[8] 2007/11/05 14:30:10: Tagging request with existing tag
[9] 2007/11/05 14:30:10: Resolve destination 6406: a udp 192.168.1.103 5060
[9] 2007/11/05 14:30:10: Resolve destination 6406: udp 192.168.1.103 5060
[7] 2007/11/05 14:30:10: SIP Tx udp:192.168.1.103:5060:
[7] 2007/11/05 14:30:10: SIP Rx udp:192.168.1.103:5060:
A delay in calls coming through
in Embedded
Posted
I've finally gotten the CS410 connected the way we wanted it and it is working. However we've noticed a delay in the calls coming through from when we had the analog phones connected. Is this normal behavior or is there something I can change?