[asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?
Hi all, I tried searching, so if this has already been discussed please point me in the right direction. On separate occasions I've seen cases where Asterisk boxes will be unable to register with each other via SIP or IAX2 when a Firewall is replaced. I'll describe two different cases. In both we have three offices connected via IPsec tunnels. Case 1: High Availability firewall fail-over We have two Palo Alto Networks PA-4020 firewalls in one office setup in an active/passive pair. Sessions and traffic are automatically maintained and moved to the passive firewall in case the active one dies/loses power/etc. When I was doing routine maintenance and had to fail over to the passive firewall purposely, the SIP connections between offices broke, and failed to re-register. What I see is: [Jun 17 10:09:40] NOTICE[3311]: chan_sip.c:7783 sip_reg_timeout:-- Registration for 'portl...@10.xx.x.25' timed out, trying again (Attempt #2273) And similarly on the other side: [Jun 17 10:09:16] NOTICE[17102]: chan_sip.c:10169 sip_reg_timeout:-- Registration for 'paloa...@10.xx.x.10' timed out, trying again (Attempt #1660) Restarting Asterisk and even both servers doesn't seem to change anything. The last time this happened, for some reason setting srvlookup=yes in the [general] section of sip.conf *seemed* to fixed it. The previous time this occured, the servers were trunked via IAX2 instead of SIP, but I switched to SIP trunks because it solved the problem (for the meantime anyway). Case 2: Entire firewall replacement In one office I recently replaced a Cisco ASA 5505 with a Palo Alto Networks PA-2020. This completely broke SIP connections to the two other offices. Same errors as above. Again, restarting Asterisk and even the servers sees no change. It seems as if somewhere there's something that is cached with regards to the old firewall (or perhaps IPsec/IKE session). I've been digging around but can't find anything obvious. Has anyone else seen this behavior and potentially found a fix? This happens with Asterisk 1.6.1.6 and Asterisk 1.4.26.2. Much thanks. - Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?
And slight update: With regards to Case 2, which happened last night. After I noticed that SIP registrations were failing between two of the offices, I commented out the register line in sip.conf on each box, reloaded SIP, and called it good for the night. After re-enabling it and reloading SIP this morning they successfully re-registered. Is there some sort of TTL, cache, saved salt value, or other time/session related tidbit saved that is expiring here? - Chris On Jun 17, 2010, at 10:21 AM, Chris Brentano wrote: Hi all, I tried searching, so if this has already been discussed please point me in the right direction. On separate occasions I've seen cases where Asterisk boxes will be unable to register with each other via SIP or IAX2 when a Firewall is replaced. I'll describe two different cases. In both we have three offices connected via IPsec tunnels. Case 1: High Availability firewall fail-over We have two Palo Alto Networks PA-4020 firewalls in one office setup in an active/passive pair. Sessions and traffic are automatically maintained and moved to the passive firewall in case the active one dies/loses power/etc. When I was doing routine maintenance and had to fail over to the passive firewall purposely, the SIP connections between offices broke, and failed to re-register. What I see is: [Jun 17 10:09:40] NOTICE[3311]: chan_sip.c:7783 sip_reg_timeout:-- Registration for 'portl...@10.xx.x.25' timed out, trying again (Attempt #2273) And similarly on the other side: [Jun 17 10:09:16] NOTICE[17102]: chan_sip.c:10169 sip_reg_timeout:-- Registration for 'paloa...@10.xx.x.10' timed out, trying again (Attempt #1660) Restarting Asterisk and even both servers doesn't seem to change anything. The last time this happened, for some reason setting srvlookup=yes in the [general] section of sip.conf *seemed* to fixed it. The previous time this occured, the servers were trunked via IAX2 instead of SIP, but I switched to SIP trunks because it solved the problem (for the meantime anyway). Case 2: Entire firewall replacement In one office I recently replaced a Cisco ASA 5505 with a Palo Alto Networks PA-2020. This completely broke SIP connections to the two other offices. Same errors as above. Again, restarting Asterisk and even the servers sees no change. It seems as if somewhere there's something that is cached with regards to the old firewall (or perhaps IPsec/IKE session). I've been digging around but can't find anything obvious. Has anyone else seen this behavior and potentially found a fix? This happens with Asterisk 1.6.1.6 and Asterisk 1.4.26.2. Much thanks. - Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?
I have a suspicion that it's the saved/cached SIP/IAX2 useragent info: -- Registered SIP 'paloalto' at 10.XX.X.25 port 5060 Saved useragent Asterisk PBX 1.6.1.6 for peer paloalto Is there a way to clear this saved info manually? - Chris On Jun 17, 2010, at 10:29 AM, Chris Brentano wrote: And slight update: With regards to Case 2, which happened last night. After I noticed that SIP registrations were failing between two of the offices, I commented out the register line in sip.conf on each box, reloaded SIP, and called it good for the night. After re-enabling it and reloading SIP this morning they successfully re-registered. Is there some sort of TTL, cache, saved salt value, or other time/session related tidbit saved that is expiring here? - Chris On Jun 17, 2010, at 10:21 AM, Chris Brentano wrote: Hi all, I tried searching, so if this has already been discussed please point me in the right direction. On separate occasions I've seen cases where Asterisk boxes will be unable to register with each other via SIP or IAX2 when a Firewall is replaced. I'll describe two different cases. In both we have three offices connected via IPsec tunnels. Case 1: High Availability firewall fail-over We have two Palo Alto Networks PA-4020 firewalls in one office setup in an active/passive pair. Sessions and traffic are automatically maintained and moved to the passive firewall in case the active one dies/loses power/etc. When I was doing routine maintenance and had to fail over to the passive firewall purposely, the SIP connections between offices broke, and failed to re-register. What I see is: [Jun 17 10:09:40] NOTICE[3311]: chan_sip.c:7783 sip_reg_timeout:-- Registration for 'portl...@10.xx.x.25' timed out, trying again (Attempt #2273) And similarly on the other side: [Jun 17 10:09:16] NOTICE[17102]: chan_sip.c:10169 sip_reg_timeout:-- Registration for 'paloa...@10.xx.x.10' timed out, trying again (Attempt #1660) Restarting Asterisk and even both servers doesn't seem to change anything. The last time this happened, for some reason setting srvlookup=yes in the [general] section of sip.conf *seemed* to fixed it. The previous time this occured, the servers were trunked via IAX2 instead of SIP, but I switched to SIP trunks because it solved the problem (for the meantime anyway). Case 2: Entire firewall replacement In one office I recently replaced a Cisco ASA 5505 with a Palo Alto Networks PA-2020. This completely broke SIP connections to the two other offices. Same errors as above. Again, restarting Asterisk and even the servers sees no change. It seems as if somewhere there's something that is cached with regards to the old firewall (or perhaps IPsec/IKE session). I've been digging around but can't find anything obvious. Has anyone else seen this behavior and potentially found a fix? This happens with Asterisk 1.6.1.6 and Asterisk 1.4.26.2. Much thanks. - Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (SOLVED) Kernel panic w/ DAHDI 2.x/Digium TE220B
FYI, in case anyone else encouters this issue. The card that I had which I could reproduce this with was hardware revision B4. I RMAed the card with Digium support and got a newer, revision C card, and the issue is no more. On 20 Oct, 2009, at 3:25 PM, Chris Brentano wrote: I've seen this consistently on three systems, with three different cards, and multiple versions of DAHDI. At first I thought the issue only occurred on newer, Nehalem-based, systems, but I reproduced it on a Core 2 Duo box as well. I've tested with dahdi-linux 2.2.0.2, dadhi- linux-complete 2.0.0+2.0.0, 2.1.0.2+2.1.0.2, and 2.2.0.2+2.2.0. The card is a Digium TE220B which uses the wct4xxp module. This does not happen, on the same systems and kernel version, with a TE121 using the wcte12xp module nor does it happen with a T100P using wct1xxp. OS is CentOS 5.3, and happens with kernel versions 2.6.18-164.el5 and 2.6.18-128.el5. I'm posting this wondering if anyone else has seen similar behavior. /etc/dahdi/system.conf: span=1,1,0,esf,b8xs bchan=1-23 dchan=24 loadzone=us defaultzone=us /etc/dahdi/modules: wct4xxp wcte12xp wct1xxp --- When I start dahdi, I see the following: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] Running dahdi_cfg: VPM400: Not Present VPM450: Not Present [ OK ] Syslog output: Oct 20 15:20:54 redbox-ast16 kernel: dahdi: Telephony Interface Registered on major 196 Oct 20 15:20:54 redbox-ast16 kernel: dahdi: Version: 2.2.0.2 Oct 20 15:20:54 redbox-ast16 kernel: ACPI: PCI Interrupt :03:08.0[A] - GSI 16 (level, low) - IRQ 169 Oct 20 15:20:54 redbox-ast16 kernel: Found TE2XXP at base address dfbfff80, remapped to c2022f80 Oct 20 15:20:54 redbox-ast16 kernel: TE2XXP version c01a016c, burst ON Oct 20 15:20:54 redbox-ast16 kernel: Octasic optimized! Oct 20 15:20:54 redbox-ast16 kernel: FALC version: 0005, Board ID: 00 Oct 20 15:20:54 redbox-ast16 kernel: Reg 0: 0x056af400 Oct 20 15:20:54 redbox-ast16 kernel: Reg 1: 0x056af000 Oct 20 15:20:54 redbox-ast16 kernel: Reg 2: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 3: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 4: 0xff01 Oct 20 15:20:54 redbox-ast16 kernel: Reg 5: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 6: 0xc01a016c Oct 20 15:20:54 redbox-ast16 kernel: Reg 7: 0x1000 Oct 20 15:20:54 redbox-ast16 kernel: Reg 8: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 9: 0x00ff00ff Oct 20 15:20:54 redbox-ast16 kernel: Reg 10: 0x004a Oct 20 15:20:54 redbox-ast16 kernel: Found a Wildcard: Wildcard TE220 (4th Gen) Oct 20 15:20:54 redbox-ast16 kernel: TE2XXP: Launching card: 0 Oct 20 15:20:54 redbox-ast16 kernel: TE2XXP: Setting up global serial parameters Oct 20 15:20:55 redbox-ast16 kernel: About to enter spanconfig! Oct 20 15:20:55 redbox-ast16 kernel: Done with spanconfig! Oct 20 15:20:55 redbox-ast16 kernel: dahdi: Registered tone zone 0 (United States / North America) Oct 20 15:20:55 redbox-ast16 kernel: About to enter startup! Oct 20 15:20:55 redbox-ast16 kernel: TE2XXP: Span 1 configured for ESF/B8ZS Oct 20 15:20:55 redbox-ast16 kernel: wct2xxp: Setting yellow alarm on span 1 Oct 20 15:20:55 redbox-ast16 kernel: timing source auto card 0! Oct 20 15:20:55 redbox-ast16 kernel: SPAN 1: Primary Sync Source Oct 20 15:20:55 redbox-ast16 kernel: VPM400: Not Present Oct 20 15:20:55 redbox-ast16 kernel: VPM450: Not Present Oct 20 15:20:55 redbox-ast16 kernel: Completed startup! --- Now if I either start asterisk, or if I stop dahdi, it will panic: # /etc/init.d/dahdi stop Unloading DAHDI hardware modules: TE4XXP: Version Syncronization Error! TE4XXP: Version Syncronization Error! TE4XXP: Version Syncronization Error! TE4XXP: Version Syncronization Error! HARDWARE ERROR CPU 1: Machine Check Exception: 4 Bank 8: 00 TSC 0 This is not a software problem! Run through mcelog --ascii to decode and contact your hardware vendor Kernel panic - not syncing: Uncorrected machine check Syslog output (not much before restart): Oct 20 07:11:54 localhost kernel: TE4XXP: Version Synchronization Error! Oct 20 07:14:24 localhost syslogd 1.4.1: restart. ... --- I only see the machine check exception on the two Nehalem boxes (HP ProLiant ML350 G6, Z800 Workstation); on a Core 2 Duo (Dell Optiplex 745) it just hard freezes after the Version Syncronization Error! messages. If there's any further details I can provide I'm happy to do so. Would like to figure out what's happening here if anyone can help shed any light as this is completely holding up migration to Asterisk 1.6 and DAHDI. Thanks. - Chris
[asterisk-users] Kernel panic w/ DAHDI 2.x/Digium TE220B
I've seen this consistently on three systems, with three different cards, and multiple versions of DAHDI. At first I thought the issue only occurred on newer, Nehalem-based, systems, but I reproduced it on a Core 2 Duo box as well. I've tested with dahdi-linux 2.2.0.2, dadhi- linux-complete 2.0.0+2.0.0, 2.1.0.2+2.1.0.2, and 2.2.0.2+2.2.0. The card is a Digium TE220B which uses the wct4xxp module. This does not happen, on the same systems and kernel version, with a TE121 using the wcte12xp module nor does it happen with a T100P using wct1xxp. OS is CentOS 5.3, and happens with kernel versions 2.6.18-164.el5 and 2.6.18-128.el5. I'm posting this wondering if anyone else has seen similar behavior. /etc/dahdi/system.conf: span=1,1,0,esf,b8xs bchan=1-23 dchan=24 loadzone=us defaultzone=us /etc/dahdi/modules: wct4xxp wcte12xp wct1xxp --- When I start dahdi, I see the following: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] Running dahdi_cfg: VPM400: Not Present VPM450: Not Present [ OK ] Syslog output: Oct 20 15:20:54 redbox-ast16 kernel: dahdi: Telephony Interface Registered on major 196 Oct 20 15:20:54 redbox-ast16 kernel: dahdi: Version: 2.2.0.2 Oct 20 15:20:54 redbox-ast16 kernel: ACPI: PCI Interrupt :03:08.0[A] - GSI 16 (level, low) - IRQ 169 Oct 20 15:20:54 redbox-ast16 kernel: Found TE2XXP at base address dfbfff80, remapped to c2022f80 Oct 20 15:20:54 redbox-ast16 kernel: TE2XXP version c01a016c, burst ON Oct 20 15:20:54 redbox-ast16 kernel: Octasic optimized! Oct 20 15:20:54 redbox-ast16 kernel: FALC version: 0005, Board ID: 00 Oct 20 15:20:54 redbox-ast16 kernel: Reg 0: 0x056af400 Oct 20 15:20:54 redbox-ast16 kernel: Reg 1: 0x056af000 Oct 20 15:20:54 redbox-ast16 kernel: Reg 2: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 3: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 4: 0xff01 Oct 20 15:20:54 redbox-ast16 kernel: Reg 5: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 6: 0xc01a016c Oct 20 15:20:54 redbox-ast16 kernel: Reg 7: 0x1000 Oct 20 15:20:54 redbox-ast16 kernel: Reg 8: 0x Oct 20 15:20:54 redbox-ast16 kernel: Reg 9: 0x00ff00ff Oct 20 15:20:54 redbox-ast16 kernel: Reg 10: 0x004a Oct 20 15:20:54 redbox-ast16 kernel: Found a Wildcard: Wildcard TE220 (4th Gen) Oct 20 15:20:54 redbox-ast16 kernel: TE2XXP: Launching card: 0 Oct 20 15:20:54 redbox-ast16 kernel: TE2XXP: Setting up global serial parameters Oct 20 15:20:55 redbox-ast16 kernel: About to enter spanconfig! Oct 20 15:20:55 redbox-ast16 kernel: Done with spanconfig! Oct 20 15:20:55 redbox-ast16 kernel: dahdi: Registered tone zone 0 (United States / North America) Oct 20 15:20:55 redbox-ast16 kernel: About to enter startup! Oct 20 15:20:55 redbox-ast16 kernel: TE2XXP: Span 1 configured for ESF/B8ZS Oct 20 15:20:55 redbox-ast16 kernel: wct2xxp: Setting yellow alarm on span 1 Oct 20 15:20:55 redbox-ast16 kernel: timing source auto card 0! Oct 20 15:20:55 redbox-ast16 kernel: SPAN 1: Primary Sync Source Oct 20 15:20:55 redbox-ast16 kernel: VPM400: Not Present Oct 20 15:20:55 redbox-ast16 kernel: VPM450: Not Present Oct 20 15:20:55 redbox-ast16 kernel: Completed startup! --- Now if I either start asterisk, or if I stop dahdi, it will panic: # /etc/init.d/dahdi stop Unloading DAHDI hardware modules: TE4XXP: Version Syncronization Error! TE4XXP: Version Syncronization Error! TE4XXP: Version Syncronization Error! TE4XXP: Version Syncronization Error! HARDWARE ERROR CPU 1: Machine Check Exception: 4 Bank 8: 00 TSC 0 This is not a software problem! Run through mcelog --ascii to decode and contact your hardware vendor Kernel panic - not syncing: Uncorrected machine check Syslog output (not much before restart): Oct 20 07:11:54 localhost kernel: TE4XXP: Version Synchronization Error! Oct 20 07:14:24 localhost syslogd 1.4.1: restart. ... --- I only see the machine check exception on the two Nehalem boxes (HP ProLiant ML350 G6, Z800 Workstation); on a Core 2 Duo (Dell Optiplex 745) it just hard freezes after the Version Syncronization Error! messages. If there's any further details I can provide I'm happy to do so. Would like to figure out what's happening here if anyone can help shed any light as this is completely holding up migration to Asterisk 1.6 and DAHDI. Thanks. - Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Whither asterisk-addons?
I noticed that asterisk.org got a redesign, quite recently it seems, which is very nice, but the addons package isn't listed for download any longer, nor are releases posted to http://downloads.asterisk.org/pub/telephony/ . That said, looks like it's still available in svn, http://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/ . So just wondering if addons will be around for the forseeable future? - Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nehalem Digium Wildcard issues?
Just putting this out there to see if anyone else has seen any issues. May cross-post to asterisk-dev if it's indeed a bug (and not my own stupidity). I've got a Digium TE220 (2xT1 interface w/Echo canceller) that in two separate Nehalem-based (Xeon E5520 Gainestown) boxes (HP ProLiant ML350 G6; HP Z800 Workstation) has caused numerous kernel panics. This is only when the dahdi service is running with a very simple config (I've defined the first span, the bchans and the dchan, and that's about it). If dahdi is stopped, or the card is removed, everything's fine. I instead installed a Digium TE122P in the ML350 and haven't had any issues. I also haven't seen this in a pre-Nehalem Xeon server. I'm using Asterisk 1.6.1.6, Dahdi 2.2.0 and LibPRI 1.4.10.1, running on CentOS 5.3 (2.6.18-164.el5). Has anyone seen anything similar? - Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whither asterisk-addons?
Correction, I did notice it for download at http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.1.1.tar.gz - Chris On 16 Oct, 2009, at 4:06 PM, Chris Brentano wrote: I noticed that asterisk.org got a redesign, quite recently it seems, which is very nice, but the addons package isn't listed for download any longer, nor are releases posted to http://downloads.asterisk.org/pub/telephony/ That said, looks like it's still available in svn, http://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/ So just wondering if addons will be around for the forseeable future? - Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy
Use ExtenSpy for spying on a specific extension. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExtenSpy On 9 Oct, 2009, at 10:44 AM, Torintino T wrote: How can i activate ChanSpy to spy on a dedicated extension? I see the following in /etc/asterisk/extensions_additional.conf [chanspy] include = chanspy-custom exten = 501**,1,Chanspy(801) exten = 501**,n,Hangup exten = 502**,1,Chanspy(802) exten = 502**,n,Hangup But when i try to call 501**, it doesn't give any response. Thanks. Torintino Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. ATT1.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-security] Person Trying to Register on my Asterisk multiple times
Hi Chris, I'd restrict access to the Asterisk box using iptables (or similar firewall) and only allow access from trusted client IPs or networks. This only works though if you know the originating IPs (and/or networks) of client connections and that they don't change over time. Alternately you could require a VPN connection between the network your Asterisk box is on and clients you anticipate connecting to it. This creates some network overhead and could introduce some latency, but is a possibility. Lastly you could block the originating IPs of attacking systems using an ACL or iptables rule, but that can quickly becoming a losing strategy if the attacker has access to different systems or different networks. Good luck! - Chris --- Chris Brentano IT Engineer Jive Software 915 SW Stark St, Suite 400 Portland, Oregon 97205 Email/XMPP: chris.brent...@jivesoftware.com On 23 Jan, 2009, at 1:36 PM, Christopher Gray wrote: Hello: Beginning on January 6, it appears that somebody has been trying to hack into my Asterisk. They have tried on the 7th, 9th, and the 20th. The messages file in /var/log/Asterisk shows entries like this: [Jan 20 13:39:40] NOTICE[5130] chan_sip.c: Registration from '1072963462sip:1072963...@198.144.206.28' failed for '212.174.78.60' - No matching peer found [Jan 20 13:39:41] NOTICE[5130] chan_sip.c: Registration from '100sip:1...@198.144.206.28' failed for '212.174.78.60' - No matching peer found [Jan 20 13:39:41] NOTICE[5130] chan_sip.c: Registration from '101sip:1...@198.144.206.28' failed for '212.174.78.60' - No matching peer found [Jan 20 13:39:41] NOTICE[5130] chan_sip.c: Registration from '102sip:1...@198.144.206.28' failed for '212.174.78.60' - No matching peer found [Jan 20 13:39:41] NOTICE[5130] chan_sip.c: Registration from '103sip:1...@198.144.206.28' failed for '212.174.78.60' - No matching peer found The sip:101 sip:102 and so on goes up until sip:9975. This began at 13:39:40 and ended at 13:42:51. Entries began at line 970 of the log file and ended at 8016 for a total of 7,041 occurrences. How worried should I be about this and what should I do to stop further attempts? Thanks for any advice. Chris ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-security mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-security ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-security mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-security
Re: [asterisk-users] Digium training course
I would also disagree that the written exam is biased towards people who attended the training. I attended a Bootcamp earlier this year and thought I was fully prepared to pass the dCAP. Especially since I already had real-world Asterisk experience. But the written exam covered material that we hadn't even discussed in class, some stuff that was in the book, and other that I was totally lost on. I passed the practical with a near perfect score, but fell just short of passing the written. IMHO, the written portion needs to be re-evaluated. What I think needs to change is de-coupling the dCAP exam from the Bootcamp class. I'll likely never retake the dCAP exam since Digium doesn't offer the Bootcamp in my area (Portland) and I can't go to a local testing facility (New Horizons, et al.) and do the exam. It would cost me well beyond the $300 to take the exam after factoring in travel costs and time spent away from work. Also, the problem with the dCAP being coupled to the Bootcamp is that it gives you the false impression that the Bootcamp prepares you to pass the dCAP and that is completely *not true*. In my Bootcamp class of 9 only 4 took the dCAP. Our own instructor said it took him 3 tries to pass! If this isn't going to change, then the dCAP should be changed so that the Bootcamp *does* prepare you to pass. And similarly, Digium should then also offer less expensive (at least, less than $3K) self-study materials or online training that also offers similar training without having to be present at the Bootcamp That way someone could elect to train at their own schedule and later coordinate to drop-in on the last day of a Bootcamp session and take the dCAP. - Chris On Sep 21, 2008, at 9:15 AM, Tilghman Lesher wrote: On Thursday 18 September 2008 20:56:58 Craig Guy wrote: I felt at the time the written portion was heavily biased towards people who had done the training - in fact I would go so far as to say that it was designed specifically to discriminate against people who had not attended the official training. I'd have to disagree with that, having taken the written portion without having attended the bootcamp, and I got one of the highest scores of the people there that day. Included was one question that I believe I was the only that day to have gotten right. Of course, I had the written the application upon which that question was based, so I had an unfair advantage, I suppose. Other than that question, though, I'd have to say that the written portion highly favored the person with a well-rounded set of experiences with Asterisk. However, the test has been revised since I have taken it, and Jared assures me that some of the more tricky questions have been removed, so the written portion may be easier nowadays. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
Actually, rethinking the original argument that the written exam is biased towards the class, I would say the exact opposite is true: The practical exam is biased towards the class. In fact, everything in the practical exam is taught in the class almost step for step. On Sep 21, 2008, at 10:20 AM, Chris Brentano wrote: I would also disagree that the written exam is biased towards people who attended the training. I attended a Bootcamp earlier this year and thought I was fully prepared to pass the dCAP. Especially since I already had real-world Asterisk experience. But the written exam covered material that we hadn't even discussed in class, some stuff that was in the book, and other that I was totally lost on. I passed the practical with a near perfect score, but fell just short of passing the written. IMHO, the written portion needs to be re- evaluated. What I think needs to change is de-coupling the dCAP exam from the Bootcamp class. I'll likely never retake the dCAP exam since Digium doesn't offer the Bootcamp in my area (Portland) and I can't go to a local testing facility (New Horizons, et al.) and do the exam. It would cost me well beyond the $300 to take the exam after factoring in travel costs and time spent away from work. Also, the problem with the dCAP being coupled to the Bootcamp is that it gives you the false impression that the Bootcamp prepares you to pass the dCAP and that is completely *not true*. In my Bootcamp class of 9 only 4 took the dCAP. Our own instructor said it took him 3 tries to pass! If this isn't going to change, then the dCAP should be changed so that the Bootcamp *does* prepare you to pass. And similarly, Digium should then also offer less expensive (at least, less than $3K) self-study materials or online training that also offers similar training without having to be present at the Bootcamp That way someone could elect to train at their own schedule and later coordinate to drop-in on the last day of a Bootcamp session and take the dCAP. - Chris On Sep 21, 2008, at 9:15 AM, Tilghman Lesher wrote: On Thursday 18 September 2008 20:56:58 Craig Guy wrote: I felt at the time the written portion was heavily biased towards people who had done the training - in fact I would go so far as to say that it was designed specifically to discriminate against people who had not attended the official training. I'd have to disagree with that, having taken the written portion without having attended the bootcamp, and I got one of the highest scores of the people there that day. Included was one question that I believe I was the only that day to have gotten right. Of course, I had the written the application upon which that question was based, so I had an unfair advantage, I suppose. Other than that question, though, I'd have to say that the written portion highly favored the person with a well-rounded set of experiences with Asterisk. However, the test has been revised since I have taken it, and Jared assures me that some of the more tricky questions have been removed, so the written portion may be easier nowadays. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)
I don't really see the advantage to be honest. If I needed Asterisk access from anywhere I'd just run it locally on my laptop, connect to an ITSP via IAX or SIP, and run a softphone app locally. The only exception I can think of is when you'd want people to be able to leave voicemail on your system when you're not connected. What are the other reasons for wanting to run it in the cloud?On Sep 9, 2008, at 7:56 PM, Steve Finkelstein wrote:Hey folks,I'm looking to potentially take some of my Asterisk servers and seehow well they fare in a cloud computing environment such as Amazon EC2+ S3. I was curious to hear feedback from anyone who's willing toshare their experience if they've already done the same. Have you hada positive experience and if not with Amazon, what other gridcomputing platform? Was it horrible and you'll never go back to it?Great ordeal of jitter/noise?Thanks a lot for your insight. :-)/sf___-- Bandwidth and Colocation Provided by http://www.api-digital.com --AstriCon 2008 - September 22 - 25 Phoenix, ArizonaRegister Now: http://www.astricon.netasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with D-channel (PRI)
I hope you have a spare card for things like this! :) If you do, I'd suggest building another Asterisk box (given you also have a spare PC for such), copy over the configs, and install the spare card. After hours, or whenever it may be appropriate, schedule some downtime and test the new system. If the same problem occurs, then you can narrow it down to the existing card or system. Then just narrow it down further. I'd probably then try the existing card in the new system, just to verify whether the card is damaged. Good luck. -Chris On 25 Aug, 2008, at 7:27 AM, Jakub Arkon Syrek wrote: We have run pattern loopback test according to http://kb.digium.com/entry/138 few times for about 3 minutes each. In one test we get 85000 errors and then there where no more errors. Zttest shows Average: 99,98% but there also comes values like 98%!! once per 10-20 minutes.. Does it mean that our card is crashed? May it be timing error on line? pri debug span 1 shows: [Aug 25 16:20:37] WARNING[2713] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! [Aug 25 16:20:38] VERBOSE[2713] logger.c: Sending Set Asynchronous Balanced Mode Extended [Aug 25 16:20:38] ERROR[2713] chan_zap.c: !! Got S-frame while link down [Aug 25 16:20:38] VERBOSE[2713] logger.c: -- Got UA from network peer Link up. [Aug 25 16:20:38] VERBOSE[2713] logger.c: q921.c:782 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED [Aug 25 16:20:38] VERBOSE[2713] logger.c: q921.c:733 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED [Aug 25 16:20:38] VERBOSE[2713] logger.c: == Primary D-Channel on span 1 up [Aug 25 16:21:07] VERBOSE[2713] logger.c: -- Timeout occured, restarting PRI [Aug 25 16:21:07] VERBOSE[2713] logger.c: q921.c:437 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED [Aug 25 16:21:07] VERBOSE[2713] logger.c: Sending Set Asynchronous Balanced Mode Extended [Aug 25 16:21:07] VERBOSE[2713] logger.c: q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH [Aug 25 16:21:07] VERBOSE[2713] logger.c: == Primary D-Channel on span 1 down [Aug 25 16:21:07] WARNING[2713] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Regards Jakub - Original Message - From: Rob Hillis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 23, 2008 12:31 PM Subject: Re: [asterisk-users] Problems with D-channel (PRI) Jakub Arkon Syrek wrote: Hello, we have strange problem, till now everything was working fine, there where no problems with dial and answer calls. Yesterday our system crashed and we notice strange behavior. What type of event caused the box to crash? Given the fact that you've also mentioned a degraded RAID array, this is sounding very much like a power spike that may have damaged hardware. What type of card is installed in your machine? Have you spoken to the relevant support people for that card? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [CentOS] Adding new Hard disk to server with RAID-5
#1.) It should just appear as unpartitioned space to the OS. You can then partition it and add that partition to one of your LVs, and then use the LVM and ext filesystem tools to grow your existing LV and then resize the filesystem to fit. Good articles on LVM: - http://kbase.redhat.com/faq/FAQ_96_4842.shtm - http://www.linux.com/base/ldp/howto/LVM-HOWTO/extendlv.html - http://tldp.org/HOWTO/LVM-HOWTO/extendlv.html - http://www.netadmintools.com/art367.html Once you learn to use LVM to your advantage you will wonder how you ever got along without it. :) Especially when you start dealing with DAS and storage shelves, etc. - Chris On 22 Aug, 2008, at 7:35 PM, Lunix1618 wrote: Hi all, I have Dell 2950 III with RAID-5 installed and managed by hardware Raid controller, I also use LVM when install CentOS. Now I get more 03 Hard disk and I would like to add it in to the running system. My question is: 1) if new hard disks add in to the machine, I have to rebuild the RAID volume with RAID management (raid controller) and the volume will be expanded, but is that make any problem to LVM at OS level ? 2) if (1's) answer is YES, what I need to do to prevent trouble occur ? If any one exprience or know the place can help me start pls share. Sorry for the dumb question but I never did this before as a newbie Linux admin - I am 'handmade' formerWindows admin Thanks for your help. regards. ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos
Re: [asterisk-users] Asterisk broadcast to web
Off the top of my head... you could probably route the audio of a softphone (like Zoiper/X-Lite) to something like Nicecast (Mac) or Icecast. On 11 Aug, 2008, at 9:21 AM, Andrew Niemantsverdriet wrote: Hi all, I have an interesting problem that I am looking for a solution for. I want to be able to call into an asterisk server and have what I say be broatcast over a streaming web radio station. I imagine using something like icecast for that. Does anybody have any pointers on how to get started? I am stuck on how to get the audio out of asterisk to be able to put into something like icecast. Any help or suggestions would be appreciated. Thanks, _ /-\ ndrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max amount of concurrent calls on and iax trunk
I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a 10Mbit link. We never did any stress testing as it's a temporary arrangement, but we've never had any call quality issues or run up against concurrent call limitations. I'm mostly routing internal extensions over the trunk, and in the case of two floating users I have their extensions at each office ring when their DID is called. One server is an older Pentium 4 1.7 GHz with 1GB Ram, and the other is a Dual Xeon 2.33 GHz with 4GB Ram. As for codec, I'm disallowing all except ulaw and gsm, with ulaw the priority codec for hardphones (Polycom) and gsm the priority for softphones (X-Lite, Zoiper). I would expect the limitation you're going to run up against is not Asterisk, but the bandwidth between your two systems. On 6 Aug, 2008, at 10:40 AM, Rosli Sukri wrote: hi, wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes connected via iax trunk. have u guys ever stress tested the trunks i.e how many concurrent calls can a trunk handle and whether codec has any effect on it. ATT1.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: LDAP auth instructions/how to?
Thanks everyone for your assistance! Just LDAP, no SSL at the moment. I configured my conf/security.properties file like so: ldap.user.store.enabled=true ldap.bind.authenticator.enabled=true ldap.config.hostname=dc02.jiveville.com ldap.config.port=389 ldap.config.base.dn=ou=JiveUsers,ou=jiveville,ou=com ldap.config.context.factory=com.sun.jndi.ldap.LdapCtxFactory ldap .config .bind .dn=cn=ldapUser,ou=ResourceAccounts,ou=JiveUsers,ou=jiveville,ou=com ldap.config.password= But cannot log in with any LDAP accounts. But I do have a couple questions: - Is there any way to test that Archiva is able to successfully talk to the LDAP server? - Are there any options above that I may be missing or which are incorrect? - When LDAP authentication is working, do all accounts that fall under the base dn OU have access? If so, what level? - Do I need to do anything in User Administrator to grant specific LDAP accounts access privileges? Thanks again! - Chris On 5 Aug, 2008, at 8:38 AM, Emmanuel Venisse wrote: I'm not sure ldap docs on redback site are up-to-date Chris, do you use LDAP or LDAPS? LDAPS isn't supported for the moment Emmanuel On Tue, Aug 5, 2008 at 5:08 AM, Maria Odea Ching [EMAIL PROTECTED] wrote: Hi Chris, You just need to put the LDAP config in your security.properties file, you no longer need to edit the application.xml as specified here: http://redback.codehaus.org/integration/ldap.html (just copy paste the config specifed in the security.properties section) And you might also need to add the LDAP specific configuration specified in the LDAP Settings section in this document: http://redback.codehaus.org/configuration.html HTH, Deng On Tue, Aug 5, 2008 at 8:16 AM, Chris Brentano [EMAIL PROTECTED] wrote: Hi all, I'd like to configure Archiva to do LDAP authentication to Active Directory. It appears that Redback has LDAP support, and I've seen some various bits here and there about configuring the security.properties or application.xml file to utilize LDAP, but I can't find a concise guide. Can anyone provide some basic instructions and are there any gotchas I should be aware of? Thanks! - Chris
Re: [CentOS] Whole disk encryption
I think TrueCrypt (www.truecrypt.org) will do this. On 4 Aug, 2008, at 8:51 AM, Plant, Dean wrote: Has there been any updates to support encrypting the whole disk in 5.2? If not, Is anyone doing this and can point me to some good documentation? Thanks Dean ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos
Re: [CentOS] DVI + VGA?
If they're both the same output (i.e. not individual outputs, but just different media options) then I'd say yes. But if they are distinct outputs (say for a multi-monitor setup) then one may be display #1 and the other #2. If it's integrated video on your motherboard there may be options in your BIOS, but if it's an add-on card then maybe consult the manufacturer's website. On 1 Aug, 2008, at 3:40 PM, MHR wrote: I have an LCD monitor with both VGA and DVI connectors on it, and a video card to match (both connectors). If I want to switch from the VGA (currently in use) to the DVI, do I need to do anything special other than switch wires? I didn't see anything in google that was helpful (though I may not have used a smashing search...). Thanks. mhr ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos
Re: [CentOS] Using CentOS 5 as server; best way to setup NFSv4?
I would second OpenLDAP, having used it in production at two different employers. It's always been stable and reliable. If you're restarting slapd every 15 minutes I'd take a good hard look at the problem versus just migrating away from it. On that note, we recently migrated to Active Directory from OpenLDAP, primarily because we migrated from Zimbra 4.5 to Exchange (and Exchange requires AD). It wasn't without much kicking and screaming, but in the end it was the best move for our users. The tricky part was switching Linux systems which had been authenticating reliably and smoothly to OpenLDAP to using Winbind instead (primarily because of AD group support). Even though it largely works, I would say that in a large production environment I prefer OpenLDAP for centralized authentication over AD, especially since we're a predominately Linux/ UNIX environment. - Chris On 1 Aug, 2008, at 5:47 PM, Craig White wrote: On Fri, 2008-08-01 at 17:33 -0700, nate wrote: I personally don't like LDAP(after having used it for many years now). I do use it at home, though only two of the 6 systems I have are actually using it(I also use it for mail routing but that is a legacy thing I setup 7 years ago that I haven't gotten around to migrating off of). I'm in the slow process of migrating my company's systems off of LDAP, they are using it for authentication and it's horribly unreliable and I hate that single point of failure and the complexity of setting it up and maintaining it. They have a cron script that restarts the LDAP services every 15 minutes and they restart nscd on all of the servers every hour. And still even I get complaints on occasion about not being able to login and I have to go restart nscd again or at least invalidate the nscd passwd cache (nscd -i passwd). LDAP is as stable as anything I've ever used but I have to admit that I don't use nscd anywhere because I would suspect, that is what is killing you. I stopped using nscd when I went to LDAP for that reason. It's not uncommon for my primary LDAP servers to have uptimes of over 9 months and never restarting though Red Hat made a curious choice of using sleepy-cat 4.3 on RHEL 5 which is totally not recommended by OpenLDAP developers. http://www.openldap.org/faq/data/cache/44.html I suppose if you wanted to have a stable LDAP, you would investigate with the developers of OpenLDAP. Craig ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos
Re: [asterisk-users] func_curl.so Error on load
Generally I'd agree. But it could at least more adequately notify the user, even if they are compiling on a different system than where it will be running on. It just seems that in most cases people will be compiling on the system they will be installing on. This is what they teach at the Asterisk Bootcamp, fwiw. Tzafrir Cohen wrote: It will: libcurl is not required for building Asterisk. Generally for most of the optional libraries, the confogure script of Asterisk will silently fail if they are not installed. I don't think you want to have to install snmp, unixodbc, openh323, libpri, libvpb and whatever just to get Asterisk built. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_curl.so Error on load
When I ran ./configure, which completed successfully, I noticed that it complained about the PKG_CONFIG_PATH and not being able to find libcurl: (lines omitted) ... checking for curl-config... /usr/bin/curl-config Package libcurl was not found in the pkg-config search path. Perhaps you should add the directory containing `libcurl.pc' to the PKG_CONFIG_PATH environment variable No package 'libcurl' found ... Which, was ridiculous that it finished ./configure and didn't error out on the spot, since without this small piece of the puzzle Asterisk would not run. So I just did a export PKG_CONFIG_PATH=/usr/lib/pkgconfig and reran ./configure and it was happy again. - Chris Tzafrir Cohen wrote: On Sat, Apr 19, 2008 at 11:11:53AM -0700, Chris Brentano wrote: Nevermind, I found the problem. And for the benefit of the readers of the archives: what was it? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Subscription/Presence Problem
I believe this isn't a Polycom thing, but the nature of SIP devices in general. But, that said, Polycom should start making IAX desk phones. :-) - Chris Lee, John (Sydney) wrote: DND does not do anything for me BLF-wise either (shame). Simply picking up the handset won't do, at that point the phone is giving you a dialtone but nothing is sent to the server. You actually have dial out. Try actually calling somebody, the state should change to InUse. Thanks Mike and Alexander. I tried out your suggestions and they are definitely true. It is a shame that Asterisk and Polycom do not support each other in this feature. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_curl.so Error on load
BTW, I did this and it did not work unfortunately. My /etc/ld.so.conf looks like: include ld.so.conf.d/*.conf /lib /usr/lib shrug - Chris Tzafrir Cohen wrote: On Sun, Apr 20, 2008 at 09:07:09AM -0500, Tilghman Lesher wrote: Tzafrir Cohen schrieb: On Sat, Apr 19, 2008 at 11:11:53AM -0700, Chris Brentano wrote: Nevermind, I found the problem. And for the benefit of the readers of the archives: what was it? Fair money on the prospect that he failed to put /usr/local/lib in /etc/ld.so.conf and run ldconfig. I'll take your bet: | This is on CentOS 5.1, kernel 2.6.18-53.1.14 on an i686. Both curl and | curl-devel 7.15.5-2.el5 are installed. Is there a symlink somewhere that | Asterisk is looking for? Or does it need a newer version of curl? libcurl is installed in /usr/lib . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] func_curl.so Error on load
Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up Asterisk (with -cvvv) I get an error regarding func_curl.so (lines omitted) ... == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF func_strings.so = (String handling dialplan functions) == Registered application 'ADSIProg' app_adsiprog.so = (Asterisk ADSI Programming Application) asterisk: symbol lookup error: /usr/lib/asterisk/modules/func_curl.so: undefined symbol: curl_global_init This is on CentOS 5.1, kernel 2.6.18-53.1.14 on an i686. Both curl and curl-devel 7.15.5-2.el5 are installed. Is there a symlink somewhere that Asterisk is looking for? Or does it need a newer version of curl? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_curl.so Error on load
Nevermind, I found the problem. Chris Brentano wrote: Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up Asterisk (with -cvvv) I get an error regarding func_curl.so (lines omitted) ... == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF func_strings.so = (String handling dialplan functions) == Registered application 'ADSIProg' app_adsiprog.so = (Asterisk ADSI Programming Application) asterisk: symbol lookup error: /usr/lib/asterisk/modules/func_curl.so: undefined symbol: curl_global_init This is on CentOS 5.1, kernel 2.6.18-53.1.14 on an i686. Both curl and curl-devel 7.15.5-2.el5 are installed. Is there a symlink somewhere that Asterisk is looking for? Or does it need a newer version of curl? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bandwidth required for Asterisk running on T1
Additionally Mark, a Channelized (also called Integrated) T1 offers 24 channels for voice/data, but after bit robbing (for signalling, etc) you only get around 56kbps per channel. ISDN PRI over T1 has 23 b-channels of voice/data and one d-channel for signalling, etc. PRI is preferred and most common. And of course, ISDN PRI over E1 gets 30 channels of voice/data and 2 channels for signalling. Jared Smith wrote: On Fri, 2008-04-11 at 01:18 -0700, mark morreny wrote: The T1 is 32 x 64Kbps channels ; Codec is GSM. That's incorrect... a T1 is 24 channels, and each channel is 64kbps. There are also a few extra bits for framing, which adds up to 1.544 megabits per second in each direction. The audio comes across a T1 as G.711 (not GSM as stated above), and on a T1 it's usually using ulaw companding. An E1 is 32 channels, and each channel is the same 64kbps. This adds up to 2.048 megabits per second. Again, the audio is in G.711 format, but alaw companding is typically used on an E1. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Conference phones
I'll third the Polycom units. We use SoundStation IP 4000s in our conference rooms and they work great. But yes, they are expensive. - Chris On 24 Dec, 2007, at 7:20 AM, Michael Graves wrote: On Mon, 24 Dec 2007 14:29:46 -, Chris Bagnall wrote: Greetings list, Does anyone have experience with SIP conference phones? I need to source a couple for a client, but I'm not really familiar with the market - i.e. what's available, what's decent quality, etc.. A cursory googling has led me to the Polycom Soundpoint IP4000 at around the £450 mark - any thoughts on this? If anyone knows a good Polycom wholesaler in the UK, I'd be really grateful if you could fire me some contact details off-list, or post them to the -biz list (to avoid cluttering this non-biz list). Can't help with the UK contacts, but the Polycom conference phones are first rate. Very pricey, but high quality devices. Dedicated conference speakerphones have huge advantages over using even a good phone with a speakerphone feature. The directional positioning of microphones and ability to add extra mics at the table ends can be a huge improvement. I've also used the Phoenix Audio Duet, which is a USB speakerphone devie for use with a soft phone client. This was a very good speakerphone device and only around $150. They have a version called the Duet Executive that allows several to be connected to provide coverage of a large board room. I see that they also have a new model called a Quattro with further enhanced capabilities, including an analog line interface. If cost is a factor I'd look at these as an alternative. Michael Graves -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [CentOS] system hibernating?
Hmm, I'm never encountered this myself. Could it be BIOS power management settings? - Chris On 24 Dec, 2007, at 8:58 AM, Jeffrey Ross wrote: I'm in the process of setting up a new system and I have found that the system is hibernating when its sitting idle for a long period of time. How do I stop this? TIA, Jeff ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos ___ CentOS mailing list CentOS@centos.org http://lists.centos.org/mailman/listinfo/centos
