03-14-2012 04:49 PM - edited 03-21-2019 05:30 AM
Hi I have been running my UC320 with out fault for a while but the call forwarding fuction has stopped working correctly
If I put call forwarding on to call my mobile number and then call the DID Sip line number for that extention it drops the call with out ringing. but all other DID's etc continue to function correcly. it only call forwards if you dial it from another customer who has the same SIP provider but from a BT line it's dead..... I have logged the issue with VoIP Unlimited to see if they can give me any ideas but it has left me stumpt. as I have re built the box three times tonight.
wee bit of info on my setup
VoIP Unlimited SIp Trunks and Secure ADSL VoIP QOS Service.
UC320W Firmware 2.2.1
Zyxel P600 Router
System is used for PBX Only on PC's on the network.
PBX run's in Blended Mode has 1x pot's Line as Backup.
4 Cisco SPA 504G Handsets
1 Fax connected to the FXS Port.
03-15-2012 06:58 AM
If I were guessing, I would say that this call is being blocked on VoIP Unlimited's side. Some SIP providers do not allow calls from CID's that are not registered with their network. I believe, by default, that the CID of the land line/cell phone gets passed through when forwarding, thus the CID coming from you to VoIP Unlimited would be the CID of the caller not your CID. This is why a call forward coming from the other person on their service works and the other does not.
03-15-2012 10:50 AM
Hi Brad,
I did think of this but there isn't any way to stop the UC320W from using the CLID from the incomming call being forwarded.or none that I can find.
I have logged the issue with the Help Desk at Cisco and have checked the Sip Trace but your Correct it's due to the VoIP Provider not allowing the number to be spoofed (rightly so) and then the call fail's I am hoping that they find a fix to this soon as my customer is going nut's over the amount of issues we have had with the box. and is not wanting to replace it with anything that works....
I have ad issues with routers Draytek ADSL Modem (not router) and others. I have today ordered the Cisco SRP-527w as that is what the cisco guys recommend for it.
Anyway thanks for the pointers much appreachiated.
Kindest Regards
Alan
03-15-2012 10:58 AM
Alan,
I kind of figured that was the issue. We don't block that kind of traffic for that very reason. We allow this because we had so many complaints similar to whatyou're experiencing. We do, however, charge a small fee for passing what is considered and invalid ANI to us. That's a passthrough charge from the upstream carriers. Most customer's don't even seem to notice or care about the charge.
03-15-2012 11:39 AM
Hi Brad,
Good that it can be fixed quickly I will asked my Account Manager regading this in the morning.
Here the Log from the SIp Trace from the VoIP Unlimiteds end, I have Highlighted the Section near the top that say's
Remote-Party-ID: "Customers Office" <sip:01xxxxxxxxx@sip.voip-unlimited.net>;screen=yes;party=calling
[Expert Info (Note/Undecoded): Unrecognised SIP header (Remote-Party-ID)]
[Message: Unrecognised SIP header (Remote-Party-ID)]
and Iam hoping that is the issue, it was ok up to a week ago and suddenly stopped working. weired. I have attached the rest of the Sip Trace below the UC320 logs .for Ref.
On the UC320 the Log shows as below
User-Agent: Cisco/UC320W-2.2.1(2)
Allow-Events: talk, hold, conference, x-spa-cti
Content-Length: 0
Mar 15 10:40:38 UC320W user.debug voice: SIP/2.0 501 Method Not Supported Here
1 (1334 bytes on wire, 1334 bytes captured)
Arrival Time: Mar 15, 2012 10:41:21.597981000
Internet Protocol, Src: xxx.xxx.xxx.xxx (xxx.xxx.xxx.xxx), Dst: 91.151.2.130 (91.151.2.130)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:079xxxxxxxx@sip.voip-unlimited.net SIP/2.0
Method: INVITE
Request-URI: sip:079xxxxxxxx@sip.voip-unlimited.net
Request-URI User Part: 079xxxxxxxx
Request-URI Host Part: sip.voip-unlimited.net
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-b4b6381e
Transport: UDP
Sent-by Address: xxx.xxx.xxx.xxx
Sent-by port: 5060
Branch: z9hG4bK-b4b6381e
From: "Customers Office" <sip:01xxxxxxxxx@sip.voip-unlimited.net>;tag=1183b2e38edb3c9o5;ref=102
SIP Display info: "Customers office"
SIP from address: sip:01xxxxxxxxx@sip.voip-unlimited.net
SIP from address User Part: 01xxxxxxxxx
SIP from address Host Part: sip.voip-unlimited.net
SIP tag: 1183b2e38edb3c9o5
To: <sip:079xxxxxxxx@sip.voip-unlimited.net>
SIP to address: sip:079xxxxxxxx@sip.voip-unlimited.net
SIP to address User Part: 079xxxxxxxx
SIP to address Host Part: sip.voip-unlimited.net
Remote-Party-ID: "Customers Office" <sip:01xxxxxxxxx@sip.voip-unlimited.net>;screen=yes;party=calling
[Expert Info (Note/Undecoded): Unrecognised SIP header (Remote-Party-ID)]
[Message: Unrecognised SIP header (Remote-Party-ID)]
[Severity level: Note]
[Group: Undecoded]
Call-ID: 8c46ec6b-bd0487e1@192.168.1.34
CSeq: 101 INVITE
Sequence Number: 101
Method: INVITE
Max-Forwards: 70
Contact: "Customers Office" <sip:01xxxxxxxxx@xxx.xxx.xxx.xxx:5060;ref=01698312090>
Contact Binding: "Customers Office" <sip:01xxxxxxxxx@xxx.xxx.xxx.xxx:5060;ref=01698312090>
URI: "Customers Office" <sip:01xxxxxxxxx@xxx.xxx.xxx.xxx:5060;ref=01698312090>
SIP Display info: "Customers Office"
SIP contact address: sip:01698312090@91.151.10.135:5060
Expires: 240
Referred-By:
Diversion: <sip:01xxxxxxxxx@sip.voip-unlimited.net>;reason=unconditional
[Expert Info (Note/Undecoded): Unrecognised SIP header (Diversion)]
[Message: Unrecognised SIP header (Diversion)]
[Severity level: Note]
[Group: Undecoded]
User-Agent: Cisco/UC320W-2.2.1(2)
Allow-Events: talk, hold, conference, x-spa-cti
Content-Length: 427
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): proxy 6020427 0 IN IP4 10.1.1.1
Owner Username: proxy
Session ID: 6020427
Session Version: 0
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 10.1.1.1
Session Name (s): sip call
Connection Information (c): IN IP4 xxx.xxx.xxx.xxx
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: xxx.xxx.xxx.xxx
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 16428 RTP/AVP 8 18 0 101
Media Type: audio
Media Port: 16428
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMA
Media Format: ITU-T G.729
Media Format: ITU-T G.711 PCMU
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-15
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-15
Media Attribute (a): sqn:0
Media Attribute Fieldname: sqn
Media Attribute Value: 0
Media Attribute (a): cdsc: 1 audio RTP/AVP 8 18 0 101
Media Attribute Fieldname: cdsc
Media Attribute Value: 1 audio RTP/AVP 8 18 0 101
Media Attribute (a): cdsc: 5 image udptl t38
Media Attribute Fieldname: cdsc
Media Attribute Value: 5 image udptl t38
Media Attribute (a): cpar: a=T38FaxVersion:0
Media Attribute Fieldname: cpar
Media Attribute Value: a=T38FaxVersion:0
Media Attribute (a): cpar: a=T38FaxRateManagement:transferredTCF
Media Attribute Fieldname: cpar
Media Attribute Value: a=T38FaxRateManagement:transferredTCF
Media Attribute (a): cpar: a=T38FaxMaxDatagram:160
Media Attribute Fieldname: cpar
Media Attribute Value: a=T38FaxMaxDatagram:160
Media Attribute (a): cpar: a=T38FaxUdpEC:t38UDPRedundancy
Media Attribute Fieldname: cpar
Media Attribute Value: a=T38FaxUdpEC:t38UDPRedundancy
Media Attribute (a): X-sqn:0
Media Attribute Fieldname: X-sqn
Media Attribute Value: 0
Media Attribute (a): X-cap: 1 image udptl t38
Media Attribute Fieldname: X-cap
Media Attribute Value: 1 image udptl t38
Frame 3 (559 bytes on wire, 559 bytes captured)
Arrival Time: Mar 15, 2012 10:41:21.598627000
Internet Protocol, Src: xxx.xxx.xxx.xxx (xxx.xxx.xxx.xxx), Dst: xxx.xxx.xxx.xxx (xxx.xxx.xxx.xxx)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 407 Proxy Authentication Required
Status-Code: 407
[Resent Packet: False]
[Request Frame: 1]
[Response Time (ms): 0]
Message Header
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-b4b6381e;rport=5060
Transport: UDP
Sent-by Address: xxx.xxx.xxx.xxx
Sent-by port: 5060
Branch: z9hG4bK-b4b6381e
RPort: 5060
From: "Your Move Bellshill" <sip:01xxxxxxxxx@sip.voip-unlimited.net>;tag=1183b2e38edb3c9o5;ref=102
SIP Display info: "Customers Office"
SIP from address: sip:01xxxxxxxxx@sip.voip-unlimited.net
SIP from address User Part: 01xxxxxxxxx
SIP from address Host Part: sip.voip-unlimited.net
SIP tag: 1183b2e38edb3c9o5
To: <sip:079xxxxxxxx@sip.voip-unlimited.net>;tag=a060b0c44d26c98b77f8f4e6919a4253.8b88
SIP to address: sip:079xxxxxxx@sip.voip-unlimited.net
SIP to address User Part: 079xxxxxxxx
SIP to address Host Part: sip.voip-unlimited.net
SIP tag: a060b0c44d26c98b77f8f4e6919a4253.8b88
Call-ID: 8c46ec6b-bd0487e1@192.168.1.34
CSeq: 101 INVITE
Sequence Number: 101
Method: INVITE
Proxy-Authenticate: Digest realm="sip.voip-unlimited.net", nonce="4f61c76e00011f402b7eddaa1524932d646ee95f4fe932bd"
Authentication Scheme: Digest
Realm: "sip.voip-unlimited.net"
Nonce Value: "4f61c76e00011f402b7eddaa1524932d646ee95f4fe932bd"
server: VOIP-UL
Content-Length: 0
Frame 18 (729 bytes on wire, 729 bytes captured)
Arrival Time: Mar 15, 2012 10:41:46.503196000
Internet Protocol, Src: xxx.xxx.xxx.xxx (xxx.xxx.xxx.xxx), Dst: 91.151.2.130 (91.151.2.130)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: CANCEL sip:079xxxxxxxx@sip.voip-unlimited.net SIP/2.0
Method: CANCEL
Request-URI: sip:079xxxxxxxx@sip.voip-unlimited.net
Request-URI User Part: 079xxxxxxxxx
Request-URI Host Part: sip.voip-unlimited.net
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-2c689fda
Transport: UDP
Sent-by Address: xxx.xxx.xxx.xxx
Sent-by port: 5060
Branch: z9hG4bK-2c689fda
From: "Customers Office" <sip:01xxxxxxxxx@sip.voip-unlimited.net>;tag=1183b2e38edb3c9o5;ref=102
SIP Display info: "Customers Office"
SIP from address: sip:01xxxxxxxxx@sip.voip-unlimited.net
SIP from address User Part: 01xxxxxxxxx
SIP from address Host Part: sip.voip-unlimited.net
SIP tag: 1183b2e38edb3c9o5
To: <sip:079xxxxxxxxx@sip.voip-unlimited.net>
SIP to address: sip:079xxxxxxxx@sip.voip-unlimited.net
SIP to address User Part: 079xxxxxxxx
SIP to address Host Part: sip.voip-unlimited.net
Call-ID: 8c46ec6b-bd0487e1@192.168.1.34
CSeq: 102 CANCEL
Sequence Number: 102
Method: CANCEL
Max-Forwards: 70
[truncated] Proxy-Authorization: Digest username="01xxxxxxxxx",realm="sip.voip-unlimited.net",nonce="4f61c76e00011f402b7eddaa1524932d646ee95f4fe932bd",uri="sip:079xxxxxxxx@sip.voip-unlimited.net",algorithm=MD5,response="3f905bf9945886add37
Authentication Scheme: Digest
Username: "01xxxxxxxxx"
Realm: "sip.voip-unlimited.net"
Nonce Value: "4f61c76e00011f402b7eddaa1524932d646ee95f4fe932bd"
Authentication URI: "sip:07912761389@sip.voip-unlimited.net"
Algorithm: MD5
Digest Authentication Response: "3f905bf9945886add37da60ba36ce341"
User-Agent: Cisco/UC320W-2.2.1(2)
Allow-Events: talk, hold, conference, x-spa-cti
Content-Length: 0
03-15-2012 09:04 AM
Hi Alan,
Thanks for reporting this issue. You might turn on logging for your SIP trunk and see if we see why the call is dropping (Status -> Support Tools -> Log). In any case suggest you open a case with the Cisco Small Business Support Center. I vaguely recall someone else with a similar problem.
Thanks,
Chris
03-15-2012 10:51 AM
Hi Chris,
I have logged the issue with the Help Desk at Cisco and have checked the Sip Trace but your Correct it's due to the VoIP Provider not allowing the number to be spoofed (rightly so) and then the call fail's I am hoping that they find a fix to this soon as my customer is going nut's over the amount of issues we have had with the box. and is not wanting to replace it with anything that works....
I have ad issues with routers Draytek ADSL Modem (not router) and others. I have today ordered the Cisco SRP-527w as that is what the cisco guys recommend for it.
Anyway thanks for the pointers much appreachiated.
Kindest Regards
Alan
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