[Asterisk-Users] FreeBSD asterisk and zaptel versions
Hello List, I seem to have a reoccuring problem trying to find the right version of the zaptel drivers to run with asterisk under FreeBSD. I had been running asterisk-0.9 with zaptel 0.7 with no problems (both built from FreeBSD ports). Yesterday I cvsup'd my ports tree and build asterisk 1.0.1_1 and zaptel 0.8_1, which seemed to work except that any attempt to play music on hold completely hangs to box. I had a similar problem in the past with moh hanging the box, and it was apparently solved by finding the magic combination of the right versions of asterisk and zaptel. I've now built the latest version of asterisk against the latest freebsd zaptel from cvs/svn, and still find the hanging problem with moh enabled. Can someone recommend a semi-recent combination of asterisk and zaptel under FreeBSD that is know to work well? I'm running 5.2-RELEASE. many thanks, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X101P FXO with RED alarm
On Mon, Jul 12, 2004 at 05:34:12PM +0100, Chris Stenton wrote: Richard, 1. don't run 0.5 zaptel driver with asterisk-head it will panic the kernel. 2. I am pretty sure that the current BSD zaptel driver only supports the fxs modules and the x100p card. Thanks Chris. My understanding was that the x100p and x101p were just the 'same thing', it didn't occur to me that there would be differences at driver level. regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X101P FXO with RED alarm
On Mon, Jul 12, 2004 at 06:29:43PM +0100, Kevin Walsh wrote: I don't really understand why people struggle with *BSD when everything just works on GNU/Linux. I see this sort of thing all the time, on lots of open source projects. Well, portability isn't really a bad thing. For me, I'm already using FreeBSD for other tasks, and would like to use it for telephony as well. best, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X101P FXO with RED alarm
On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin Walsh wrote: Richard Airlie [EMAIL PROTECTED] wrote: First things first. Scrap the ports and build from the latest CVS source. 0.9 is far to old and buggy, and suspect the same of the Zaptel driver you have, although I don't use *BSD myself. I cvsup'd to the latest source yesterday and tried to build zaptel, but it failed right away. (trying to include linux/*.h) I didn't try building asterisk as it seems like the problem is with zaptel -- i.e. I should be able to load the zaptel driver and not see a red alarm, irrespective of my asterisk version, right?. Secondly, the red alarm does tend to mean that the line is not connected, but I got what you're describing when I moved Asterisk to a new machine. Try the X100P card in a different PCI slot. That cleared it for me, for whatever reason. Thanks for that, I gave it a try but unfortunately it's made no difference. I am suspecting the problem is either with the zaptel driver in ports (which is the only version I can get to build) or i've got a hardware issue. For what it's worth I can plug a phone into the back of the FXO and get dial tone, so I guess that proves that the cabling is OK? best, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P FXO with RED alarm
Hi, I've just added an X101P FXO card to my asterisk machine. The card is detected ok but zttool always shows it in a RED alarm state, which I understand means that it doesn't detect the line. I've connected the card via the line socket to my analogue line using the cable that came with the card (and have also tried other cables too) but I always get a RED alarm. I'm located in the UK, running FreeBSD-5.2, with zaptel-0.5 drivers and asterisk-0.9.0_1 (both built from ports). My zaptel.conf: fxsks=1 loadzone=uk defaultzone=uk My zapata.conf [channels] signalling=fxs_ks context=default channel = 1 Output of ztcfg -vv: Keyword: [fxsks], Value: [1] Keyword: [loadzone], Value: [uk] Keyword: [defaultzone], Value: [uk] Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Asterisk sees the zap channel (or at least it shows up with 'zap show channels') but always sees it as busy, presumably because of the red alarm. The other thing of note is that when the zaptel driver is loaded, attempting to play MOH causes the entire system to lock up and require a reboot. (Without the driver loaded it works, but is of course choppy because of the lack of a timing source). Anyone have any ideas about what's going on? Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] freebsd?
On Wed, Apr 14, 2004 at 04:56:46PM -0700, Randy Bush wrote: the freebsd port tree version is dead because of the openh323 issues. before i start hacking, i am hoping someone else has a freebsd version that will build on -current. and i do not care about h232. Just comment out the line with FORBIDDEN= in the port Makefile. (You will need to do it for pwlib and openh323 as well, if I recall correctly). You can then make install it in the usual fashion. If you are worried about the H323 security issues then I guess you will need to do some hacking or reconfiguring to get rid of it. Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
On Mon, Apr 05, 2004 at 10:26:32PM +0100, Richard Airlie wrote: On Mon, Apr 05, 2004 at 11:16:39AM -0500, Bob Klepfer wrote: Olle E. Johansson wrote: Richard Airlie wrote: On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? Just to follow up on this, my problem was that a long time ago I'd configured the Asterisk machine, which also happens to be a firewall, to fwd packets arriving on port 5060 into my LAN. This is why Asterisk never saw any SIP traffic - it was being forwarded elsewhere by the firewall. I reconfigured the firewall to accept packets on 5060 and all is well. regards, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone registering problem
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: download ethereal and take a peek at the packets on the wire. Without something like that, no one is really going to be able to help you. Do you mean then that my SIP trace displayed at kphone looks otherwise OK -- that the REGISTER request that kphone's sending out looks alright? Is there a single good resource describing the SIP protocol specification so I know what I should be looking for? thanks, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phone registering problem
I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454 REGISTER sip:192.168.100.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.13;rport CSeq: 4399 REGISTER To: sjphone2 sip:[EMAIL PROTECTED] Expires: 900 From: sjphone2 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.0 Event: registration Allow-Events: presence Contact: myusername sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK SipClient: Sending to '192.168.100.3:5060' SipClient: Receiving message... SipClient: Received: 21:47:45.471 - REGISTER sip:192.168.100.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.13;rport CSeq: 4399 REGISTER To: sjphone2 sip:[EMAIL PROTECTED] Expires: 900 From: sjphone2 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.0 Event: registration Allow-Events: presence Contact: myusername sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK SipCall: Incoming request SipCall: New transaction created SipTransaction: Incoming Request SipTransaction: Retransmit 1 (4000) SipClient: Sending: 21:47:49.456 REGISTER sip:192.168.100.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.13;rport CSeq: 4399 REGISTER To: sjphone2 sip:[EMAIL PROTECTED] Expires: 900 From: sjphone2 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.0 Event: registration Allow-Events: presence Contact: myusername sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK SipClient: Receiving message... SipClient: Received: 21:47:49.466 - REGISTER sip:192.168.100.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.13;rport CSeq: 4399 REGISTER To: sjphone2 sip:[EMAIL PROTECTED] Expires: 900 From: sjphone2 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.0 Event: registration Allow-Events: presence Contact: myusername sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK SipCall: Incoming request SipTransaction: Incoming Request Retransmission SipTransaction: Response Retransmission SipTransaction: Retransmit 2 (4000) (and so it continues) Seems like the REGISTER messages are being recieved at Asterisk but then just echoed back to the SIP phone? What am I doing wrong? thanks! Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on FreeBSD
Hello, Asterisk in FreeBSD ports is currently FORBIDDEN due to security issues raised in pwlib (H323). As I just want to test Asterisk internally at this point I commented out the FORBIDDENs and compiled it with no problems. Unfortunately though, I can't seem to get any SIP softphones to register with Asterisk. I have tried SJPhone on Windows and KPhone and Linphone under FreeBSD. At the Asterisk console I've turned the sip debug on, but don't see anything at all. (no SIP traffic). I have followed the quickstart quide and configured things in sip.conf, but with no success. At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) The version I installed is 0.7.2, running on FreeBSD 4.7. many thanks, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: Richard Airlie wrote: At this stage, and someone please confirm that Asterisk is really working on FreeBSD ? :) Yes, it's working with some limitations. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd Thanks for that, good to know. And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? thanks, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
On Mon, Apr 05, 2004 at 11:16:39AM -0500, Bob Klepfer wrote: Olle E. Johansson wrote: Richard Airlie wrote: On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote: And now leads me to ask... why should my SIP softphones be unable to register? They are on the same subnet as asterisk. If i have sip debug turned on, shouldn't I at least be seeing some action on the Asterisk console when they try to register? Turn on SIP debug and you'll be able to see what happens. Check with sockstat -l if Asterisk is listening to port 5060. Also, make sure you start asterisk with a lot of -v to get debug output. Since you see no SIP traffic with SIP debug on, is ipfw blocking SIP? I'm actually running IPFilter, but I've checked the logs and it definitely isn't blocking any SIP traffic. And I've also confirmed that Asterisk is listening on port 5060 with netstat. So.. Asterisk is running, listening on UDP port 5060, the firewall hasn't logging any blocked packets, and yet my IP softphones still cant register. This leads me to believe I must be doing something really stupid. My Asterisk server is 192.168.100.3. Kphone is running on 192.168.100.13, and SJPhone is on 192.168.100.11. I'm configuring the softphones so that they register with the (outbound) proxy at 192.168.100.3. I've set their IDs to be sip:[EMAIL PROTECTED], and created the appropriate username and password in sip.conf on Asterisk. I turn sip debug on at the Asterisk console, then restart the phones. They log lots of attempts to register in the softphone windows, but Asterisk doesn't see anything at all. (I can also get the softphones to talk directly to one another and they seem to be working fine). I guess my next step will be tcpdump.. but any other suggestions most welcomed! best, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
