Cannot make outbound call on SPA112: Dials but then silence: Kern.err -22

[Tanya Crawford]

When I try to make an outbound call I get a kernel error -22.

The remote end doesn't ring - Just silence on the line

The service registers fine, and I can receive incoming calls

I've factory reset and reconfigured the ATA a few times

What does this mean?

Please refer to the attached log.

Thanks

[Dan Lukes]

This log is not so helpful. Not contain no debug level messages. So reconfigure logging and try again.

Other information may help us as well. Thus turn on application level syslog&debug messages (highest level possible) and catch them. Also, capture SIP datagrams exchanget with PBX.

By the way, I assume you are running latest firmware. If not, upgrade first.

[Tanya Crawford]

So there is no way to tell what this means?

Dec 5 01:51:27 SPA112 kern.err [17179679.648000] voice_kmode_thread(0): sock_sendmsg() failed witherror -22

I have a Cisco SPA112 connected to a Uniden Cordless phone on Line 1.

There is no PBX.

The Firmware is 1.4.1. That was the first thing I did when I purchased the SPA112.

The attached image shows the log settings are set to debug level, and I have a syslog server running. It's not getting any messages from the ATA

Thus turn on application level syslog&debug messages

If you mean the SIP debug option on line1 it is set to "Full".

Also, capture SIP datagrams exchanged with PBX

I don't know what that means

My question was what does the error message -22 in the log file mean? I cannot find a list of SPA112 error codes anywhere.

I cleared the log, attempted the outbound call. This is everything in the log.

[Dan Lukes]

Just guessing, but error 22 is "Invalid argument". With no additional information mentioned above I can't even guess what argument is invalid.

SIP debug option on line1 it is set to "Full"

But you didn't captured the log messages generated.

The log you provided is internal log with no value for the purpose of voice call issue analysis. I'm asking for log of voice application which is not captured by SPA112 itself.

You must run own syslog server, configure it's address in voice application config and catch the messages sent. Read here for more.

Also, you should capture all packets sent and received from/to SPA112 related to the call in question. Ask your local network administrator for help if you don't know how to capture packets passing thru your's LAN. There's no generic advice possible as it depend on your local topology and networking equipment used.

So sorry, but no way to help you unless you can

[Tanya Crawford]

Got syslog server working, but there are no debug level messages coming from the ATA. See attached XML file (in the ZIP file)

Captured packets using wireshark. Again, there are no debug level messages. See attached txt file. (In the zip file)

There is no network administrator.

Attached are ATA configurations. You can see that SIP logging is set to full, and debug level is set, but there are no debug messages being sent.

I had previously followed the instructions on the link you had sent me, but note that the screen caps in that do not match the SPA112 @firmware 1.4.1. There is no debug server option.

I don't know what else to do to get the ATA to send debug level messages.

Also.. this is the dial plan (since the screen cap was cut off I've pasted it manually)

(000S0|106S0|111S0|*xx.|183[12]x.|1[38]00xxxxxxS0|19xxxxxxxx!|04xxxxxxxxS0|13[1-9]xxxS0|0[2378]xxxxxxxxS0|[2-9]xxxxxxxS0|001xxxx.S5)

I'm asking for log of voice application which is not captured by SPA112 itself.

There is no voice application, if you mean something like a soft phone.

As I said previously, I am using a Uniden 8155 cordless phone system. It is connected to line 1 of the ATA. Also, the ATA is connected to my ASUS RT-AC87U router, which in turn is connected to UNI-D 1 on the FTTP NTD.

btw: I cut off the phone numbers in the attached images for privacy reasons - they are not missing.

What else can I tell you?

thanks

[Dan Lukes]

The syslog captured is contain just kernel syslog messages only. There's no one voice application syslog message recorded. So the logging is still not configured properly. Please read and follow the document I mentioned already. Both syslog server and debug server IPs needs to be configured. Debug level needs to be 3 (highest available).

Even if your syslog server will not record messages, they will be captured by wireshark.

Please attach native wireshark file (rename it to .pcap.txt so it will not be rejected) - neither xml not txt can be opened in wireshark and analyzed easily.

screen caps in that do not match the SPA112 @firmware 1.4.1. There is no debug server option.

Disclose the screenshot (don't attach - use "add media" button to embed picture into comment). As a blind guess I assume you are either not logged in as administrator, or you didn't switched the UI to advanced mode. Did you noticed the configuration page in question is on "Voice" tab, not the "Administration", isn't it ?

There is no network administrator.

Someone has designed the network, including the network topology, someone has configured and connected network infrastructure devices (switches, routers, ...). This person is network administrator.

ATA is connected to my ASUS RT-AC87U router, which in turn is connected to UNI-D 1 on the FTTP NTD.

So double NAT and firewall on the path ? Well, it's premature question. Voicemail syslog messages should be good start point for now.

[Tanya Crawford]

Ok, try the attached.

Syslog Watcher will only allow xml and csv export and since I cannot upload either both the xml file and the wireshark cap file are in the attached zip.

I guess I am the network administrator

[Tanya Crawford]

Well, I got the syslog and debug messages that were requested and that was a week ago..

Can't waste any more time. I've got my VSP to set up another line using their POS router. Crappy but it works.

[Dan Lukes]

No, you provided logs required three days ago not the week ago.

Although I'm sad you are not satisfied, please note this is volunteer support forum. I'm trying to help you in my free time a and had no such time in the past two days.

If there's a business loss because of the issue you are facing, you should contact vendors official support center. I assume you have active support contract.

[Tanya Crawford]

Geez, what else is there to "turn on". The SIP debug was set to full. I had syslog and debug servers configured, I gave you every packet wireshark picked up from the ATA...

Any way, No, I provided the logs a week ago. I edited the message 3 or 4 days ago and removed part of the content.

Anyway, it doesn't matter, I had to go another route.

I understand this is a volunteer forum. I was not angry. I just couldn't wait any longer to get a resolution to the issue.

Though no fault of anyone's here, the problem has existed since over two weeks.

Every time I edit the message it seems to time stamp it at the time I edit it, instead of leaving the original date/time, with the time edited at the end.

It's a bit tough on you being the only person that seems to provide support.

There is no business.

Any way, have a good Christmas and best wishes.

Thanks for at least trying.

[Dan Lukes]

Well, you go another route, so just for the completeness - the capture of SIP packet is not matter of phone's configuration. The capturing device can;t capture packets that doesn't reach it's interface. THsu capturing device needs to be connected to proper point of LAN. Which point is "proper" ? It depend on network topology and infrastructure devices used. Ask network administrator for help with capturing particular traffic.

It's a bit tough on you being the only person that seems to provide support

You mean volunteer support for free, of course. You can still buy support contract and call Cisco Small Business Support Center for help. Well, we have no good experience with it. Support from former Linksys department has been excellent. Current CSBSC support is rather "no support". We are buying support contracts no longer, but it's another story ...

Best Christmas and happy New Year to you ...

[Dan Lukes]

Yes, it's the syslog I wished for. It will help a lot.

According the log, the error has occurred after the 183 SIP progress message has been received from peer. Unfortunately,  you captured just syslog messages not the SIP, thus I know no content of SIP packet in question. Thus I can guess only. A assume the error may be related to SDP content of progress message. There's syslog message claiming remote IP/port to be 0.0.0.0:0 - which is incorrect (or very suspicious, at least).

With no content of SIP packets I can guess nothing more.

It may be caused by either buggy content of providers response, or the content is just uncommon and you hit firmware bug.  Only SIP packets captured can disclose us more.

[Tanya Crawford]

Can someone please help me with this. I am paying for a phone service that I cannot make calls from.