CNAM Error on strange Caller Id

Beta Program Issues
1

Hello,

I’m unable to accept calls from a phone number and it appears to be an issue with the CNAM that they are sending. They are sending "20 " for the CNAM. The number calling is the “Yellow Pages” in Canada. This is the only time I have ever seen this error happen in FreePBX, so I assume it is because of the back-slash. Here are the errors:

[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as187ad96c’
[2011-07-06 08:55:57] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as187ad96c’
[2011-07-06 08:55:57] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as732edfc4’
[2011-07-06 08:55:57] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as732edfc4’
[2011-07-06 08:55:57] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as689704d8’
[2011-07-06 08:55:57] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as689704d8’
[2011-07-06 08:55:57] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as505b52f9’
[2011-07-06 08:55:57] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:57] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:57] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as505b52f9’
[2011-07-06 08:55:57] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as2888eaf8’
[2011-07-06 08:55:58] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as2888eaf8’
[2011-07-06 08:55:58] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as5c0afc31’
[2011-07-06 08:55:58] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as5c0afc31’
[2011-07-06 08:55:58] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as679a56e7’
[2011-07-06 08:55:58] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as679a56e7’
[2011-07-06 08:55:58] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as0d6a1b4b’
[2011-07-06 08:55:58] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:58] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:58] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as0d6a1b4b’
[2011-07-06 08:55:58] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:59] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:59] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as1e7beb09’
[2011-07-06 08:55:59] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:59] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:59] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:59] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as1e7beb09’
[2011-07-06 08:55:59] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?
[2011-07-06 08:55:59] WARNING[2800] sip/reqresp_parser.c: No ending quote for display-name was found
[2011-07-06 08:55:59] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as014cd0b4’
[2011-07-06 08:55:59] NOTICE[2800] chan_sip.c: From address missing ‘sip:’, using it anyway
[2011-07-06 08:55:59] VERBOSE[2800] netsock2.c: == Using SIP RTP TOS bits 184
[2011-07-06 08:55:59] VERBOSE[2800] netsock2.c: == Using SIP RTP CoS mark 5
[2011-07-06 08:55:59] WARNING[2800] sip/reqresp_parser.c: No closing quote found in ‘"20 " sip:[email protected];tag=as014cd0b4’
[2011-07-06 08:55:59] WARNING[2800] chan_sip.c: Not a SIP header ("20 " <sip:8662876208)?

When the caller calls, FreePBX tries several times to decipher the SIP header.

Has anyone else ever seen this before? Do you know if it can be fixed?

Thanks,
Darrell.

2

Good day,

I’m still having issues with the CNAM from certain callers that come into my FreePBX box with certain characters in the CNAM. The past couple of days, I’ve been getting calls from a company who for some reason have a back-slash “” at the end of their name. FreePBX cannot handle this. IE:

CNAM = "DOE JOHN"
NUMB = <6139991221>

FreePBX rejects this call due to the back-slash. Looking in the log, FreePBX cannot interpret the backslash and issues a hang-up command. The VoIP company gets a signal that I’m not registered, and forwards the caller to the voip company’s failover routing.

Is there any way to strip any non-alphabetic characters out of the CNAM before it goes through the FreePBX logic and ultimately gets rejected? The situation is very easily reproduced, and fails each and every time. I’m using the production 2.9 version of FreePBX. I cannot use the 2.10 beta version because there is a small business running off of this 2.9 version. It might be risky switching to beta, but maybe the cnam logic in FreePBX has been re-geared to allow any sort of characters in the CNAM?

Thanks for any help you can offer.

Darrell.

3

I’m surprised that nobody technical has responded to this thread. Unless they do, I’d suggest that you open a bug report.

http://www.freepbx.org/trac/newticket

4

Well thanks for replying. At least it confirms that my messages are visible. :slight_smile:

I will open a bug report. Thanks.

I went through the dumps and what appears to happen when FreePBX sees the back-slash “”, it assumes its the end of the caller id string and doesn’t even read the number. It goes into some kind of error logic with the message, “Invalid SIP Header”. The calls with a \ are rejected due to this error. There’s no way for the call to get through.

5

I’ll try to bring this thread to a developer’s attention, but they’ll just say that the right way to bring things to their attention is to open a bug report. :slight_smile:

6

I have a feeling this might be an asterisk issue, it looks to me as if the call is failing right at the beginning before it hits the from-pstn context in /etc/asterisk/extensions.conf.

7

I did open a bug report.

I found an AGI to fix CNAM when there issues involving special characters. The instructions mention putting it in extensions.conf. However I can’t do that with FreePBX. So I put it in extensions_custom.conf. It did absolutely nothing. The backslash still caused the calls to terminate. The phone doesn’t even ring. The call gets kicked out like it was on the blacklist. So you may be onto something Stonet.

8

see #5439 for further info and use that ticket you opened to provide additional feedback, though this is looking very likely to be an Asterisk issue that we may not be able to control.

9

Whoops. It’s ticket number 5438.

Thanks for the heads up.

10

If this is an Asterisk bug, you need to go and open a bug report with Digium.

Follow this link:

https://issues.asterisk.org/main_page.php

And thank you for your contributions to making Asterisk and FreePBX better for everyone. :slight_smile: