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SPA112 Call waiting- Busy line?

Hello everyone


I'am having some trouble getting my call waiting to work, I can put a call on hold by taping the hook switch/plunger. That gives me a dial tone and the other phone hears call hold music, but if i try to call out it beeps busy and if I try to make a call to that phone it also beeps busy.


So i have found some settings that i have played around with without results.


Hook Flash Tx Method: None (AVT) (INFO)

Hook Flash MIME Type: application/hook-flash

Referral Services Codes: (Blank)

Hook Flash Timer Min: .1

Hook Flash Timer Max: .9

Call Waiting Serv: yes

CWCID Serv : yes

CW Setting: yes

CWCID Setting: yes

Default CWT: 1


Are there some special requirements to use call waiting with a SPA112, for example, do I need a PBX?


All suggestions deeply appreciated.


I'm quite new to ATAs and I'm no IT technician but I have used and been around computers for almost all my life so I do know some stuff, but i guess my explanations can get a bit strange as I don't fully know what I'm talking about. Also not native English speaker.




Everyone's tags (2)

Re: SPA112 Call waiting- Busy line?

Second call is rejected by either SPA112 or remote SIP proxy. So you need to identify issue cause first. Capture SIP packets and check content (ask your LAN administrator for help). If there's SIP INVITE related to the second call, but negative response from SIP proxy, then it's SIP proxy issue. If there's no SIP INVITE related to second call, it's SPA112 issue. Capture logs of Voice application for further analysis then.


Related: Debug and syslog Messages from SPA1x2 and SPA232D ATA


Re: SPA112 Call waiting- Busy line?

I'm a one man operation so i don't have any technician nor administrator to ask for help, I do everything my self.

So i attempted to capture with Wireshark but i don't think I did it right.


I opened Wireshark, selected my local network, typed in "host" and my Cisco ip address, did some test phone calls  and tried to use call hold/waiting, stopped capture and saved it to the file i included here.


Re: SPA112 Call waiting- Busy line?

I'm a one man operation so i don't have any technician nor administrator to ask for help, I do everything my self.

Plausible, but you need to learn LAN basics and how the switching works.

Just very simple introduce to switching - communication send from switch port S (for example the port where SPA112 is connected to) to switch port U (for example port to upstream Internet provider) is not copied to switch port P (for example the port where your PC is connected to).

It's why your PC see no packets between SPA112 and upstream.

I can't advise how you can capture the packets. It depends on switch capabilities, route capabilities, LAN topology ...


So i attempted to capture with Wireshark but i don't think I did it right.

I see multicasts send by phone, but no other communication. I'm pretty sure it's because switch doesn't send the packets to PC, thus thy can't be captured. Ask your local network guru for help, unless you have no enough time to learn LAN/switchng basics by self. You may try to capture SPA112 voice application syslog&debug instead. It may (or may not) be enough to analyze the issue cause. In the first case the packet capture will be no longer necessary.


Re: SPA112 Call waiting- Busy line?

So i did a new capture after enabling port mirroring on the switch with my SPA112, got a lot more stuff captured.

Still uncertain if i have done everything right top capture what is needed to diagnose the issue.


Re: SPA112 Call waiting- Busy line?

While I can provide short reply "second call gets rejected by your provider" I will describe all major events observer in the captured file instead. Texts in () are copied from captured packets - it may help you to read and understand dump by self.

OK, lets go.

I see first call setup - INVITE from 03.....16 to 17223 (Mattias). Call is ringing (new state CC_CST_RINGING), caller info has been transmitted to analog phone (CID_initGen -> CID:OnHookTx -> CID:DONE, DTMF method used to transmit), off hook detected (Off Hook, new state CC_CST_ANSWERING),  call setup completed (new state CC_CST_CONNECTED) with bidirectional audio (Going from Rx only to bi-directional). RTP packets in both directions are captured as well.

So far so good.

Now the second call. It start by flash (Hook Flash). Current call (channel 0) is moving to hold (new state CC_CST_HOLDING), audio switched off (RTP channel 0 going from Bi-dir to Rx, RTP channel 0 going from Rx to Idle), current call is now on hold (CC_CST_HOLD), new channel 1 is initialized to accept dialing (new state CC_CST_DIALING).


I see no dialing from phone. Instead of it, I see another flash (Hook flash) - just three seconds after the first one. It's resume request.


So the channel 1 stops waiting for dial (call 1: old state = CC_CST_DIALING, new state CC_CST_IDLE) and channel 0 is going to resume (call 0: old state = CC_CST_HOLD, new state CC_CST_RESUMING). Call is considered CC_CST_CONNECTED again, bidirectional audio gets restored (*) successfully (Going from Tx only to bi-directional). I see bidirectional RTP audio again.


Another Hook flash. Call 0 go in hold, call 1 initiated and waiting for dialing (no detailed description of steps, it's same as previously).


DTMF digit 0 detected (EVENT_DTMFON 0 -> EVENT_DTMFOFF 0). All digits collected, going to dial (call 1: old state = CC_CST_DIALING, new state CC_CST_CALLING, Calling:07......54). SIP INVITE has been sent to provider (see packet #3775 of dump). Such request is rejected by provider (packet #3790):

SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP;branch=z9hG4bK-23003fe;received=185.x.y.28
To: <>;tag=c8942761
From: "Mattias"<>;tag=3b2954a820e81106o0
Call-ID: e66eaed0-4ca23cfe@
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem (493)
Warning: 499 "Not available"
Content-Length: 0

Conclusion: there's nothing wrong with your configuration. Second call setup is just rejected by upstream VoIP provider. No way to identify true cause - we can just guess. It's because of provider policy. True cause can be disclosed by provider only. So sorry for so long answer claiming just "not your fault" ...

Note the PCAP file you attached contain phone numbers (calling and called) as well as recorded audio. It contain no credentials (passwords). If you consider the file content sensitive, you may remove it now.


(*) Its time of "uchConnectEpToNode(): Error Connecting EP VoIP 0 to Node 0. -93." message. Despite the phrase "Error" in text, the message is claimed INFO severity. E.g. no ERROR and even no WARNING. Based on context i assume it's attempt to initialize a RTP stream related structure. Such attempt fails because this is resume only - the structure is initialized already.  It seems the message mean nothing important for us.


This post contain information related to other's site thus it may contain information considered Sensitive or Confidential.