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Call Forwarding to an External Number Issue

cameronalan
Level 1
Level 1

‎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.

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0 Helpful
6 Replies 6

rowseyba1
Level 1
Level 1

‎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.

0 Helpful

cameronalan
Level 1
Level 1
In response to rowseyba1

‎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

0 Helpful

rowseyba1
Level 1
Level 1
In response to cameronalan

‎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.  

0 Helpful

cameronalan
Level 1
Level 1
In response to rowseyba1

‎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
0 Helpful

Christopher Edgeworth
Level 6
Level 6

‎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

0 Helpful

cameronalan
Level 1
Level 1
In response to Christopher Edgeworth

‎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

0 Helpful
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