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Unable to make internal calls


badgewick
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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:

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Internal calls do not run through the dial plan.

 

If there is a huge delay that usualy points to problem in the routing. Sometimes the SIP requests take a long way out to the Internet and then back through the firewall.

 

If you turn the SIP packet logging on you will see more about the routing of the SIP packets and where they come from. That might help to locate the problem.

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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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