| Summary: | ASTERISK-14996: CLI incorrectly delivered when there is no calling number | ||
| Reporter: | ilker Aktuna (mrmrmrmr) | Labels: | |
| Date Opened: | 2009-10-15 14:46:56 | Date Closed: | 2011-06-07 14:07:53 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Transfers |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) incoming_no_cli.cap ( 1) incoming_no_cli.txt | |
| Description: | I am using a SPA 3000 as a PSTN gateway. Incoming PSTN calls are connected to Asterisk through SPA 3000 (it has a fxo port) via SIP. Everything is fine with this call scenario, but if the incoming PSTN call has no caller ID, then Asterisk receives the call with contact header and from header as "sip:192.168.254.5" When it sends the same call to an internal extension Asterisk adds number "192168254254" as caller ID to both from and contact fields. "sip:192168254254@192.168.254.5" I checked all configuration files but couldn't find a way to remove this caller ID. These calls don't have a real calling number, so they should be delivered to the extension without any calling number. I will add the SIP trace captured on Astersik console and the network trace including the SIP packets to the issue. | ||
| Comments: | By: Elazar Broad (ebroad) 2009-10-15 15:22:42 What do you have for sdpowner in sip.conf? By: ilker Aktuna (mrmrmrmr) 2009-10-15 15:26:42 I couldn't find anything like sdpowner in any of the configuration files (including sip.conf) By: Elazar Broad (ebroad) 2009-10-15 15:28:15 Hmm, I have a hunch...can you set sdpowner to say asterisk, i.e. sdpowner=asterisk and test again? Thanks! By: ilker Aktuna (mrmrmrmr) 2009-10-15 15:34:00 didn't make any difference. By: Jared Smith (jsmith) 2009-10-16 09:36:20 I'm not sure this has any relevance, but I find this line interesting: Found peer 'pstn sip' for '192.168.254.254' from 192.168.254.5:5061 Do you really have a peer section in sip.conf with a space in it, like this? [pstn sip] If so, you may want to try removing the spaces from your peer name and see if that makes a difference. As far as I know, it's not valid to have spaces in section names in any of the Asterisk configuration files. By: ilker Aktuna (mrmrmrmr) 2009-10-16 12:46:12 ok; I changed it to pstn-sip but the problem persists. | ||