Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.
On 2017-04-18 05:21 PM, Duncan Turnbull wrote: Sent from my iPhone On 19/04/2017, at 11:43 AM, Ernie Dunbar <maill...@lightspeed.ca> wrote: On 2017-04-18 03:38 PM, Duncan Turnbull wrote: -- Original Message -- From: "Ernie Dunbar" <maill...@lightspeed.ca> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Sent: 19-Apr-17 10:25:59 AM Subject: [asterisk-users] SIP connections over OpenVPN connection get one-way voice. Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't working *quite* as well as we'd hoped. First, here's our technical details: The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT router. The router has UDP port 1194 forwarded to our server. This server also runs our office Asterisk PBX, so there isn't any networking hardware or firewall between the VPN tunnel and the Asterisk PBX. Asterisk maybe replying from the TUN address which may confuse your sip client - if you set the TUN address as a proxy that seems to solve it. If asterisk is bound to every address then implicitly it shouldn't matter where it replies from, but in the openvpn case it seems to reply from a different address to the one it was called on and that can definitely fool clients. tcpdump on the tunnel can help you see whats happening I think I'll need a bit more detail about how to set the TUN address as a proxy. Is this done on the OpenVPN server, or at the client end? I'm also going to tell Asterisk to bind to all IPs and then restart it when there's no calls in progress, perhaps that's all I need to do? Set it as a proxy server in your sip phone client, we found using the tun ip on the vpn server works, we keep the actual asterisk address as the sip server and use the tun ip as the proxy server Asterisk is probably already bound to all the addresses netstat -nupl should show you the addresses it's listening on for udp, if it says 0.0.0.0 it means all addresses sudo tcpdump -i tun0 -s0 -A udp port 5060 Should show you the sip messages going through the tunnel and you can check the reply addresses Hmm. I also can't ping the phone's IP address on the 192.168.1.0/24 network. Perhaps that's the real problem there. This VPN should work both ways, shouldn't it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.
On 2017-04-18 03:38 PM, Duncan Turnbull wrote: -- Original Message -- From: "Ernie Dunbar" <maill...@lightspeed.ca> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Sent: 19-Apr-17 10:25:59 AM Subject: [asterisk-users] SIP connections over OpenVPN connection get one-way voice. Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't working *quite* as well as we'd hoped. First, here's our technical details: The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT router. The router has UDP port 1194 forwarded to our server. This server also runs our office Asterisk PBX, so there isn't any networking hardware or firewall between the VPN tunnel and the Asterisk PBX. Asterisk maybe replying from the TUN address which may confuse your sip client - if you set the TUN address as a proxy that seems to solve it. If asterisk is bound to every address then implicitly it shouldn't matter where it replies from, but in the openvpn case it seems to reply from a different address to the one it was called on and that can definitely fool clients. tcpdump on the tunnel can help you see whats happening I think I'll need a bit more detail about how to set the TUN address as a proxy. Is this done on the OpenVPN server, or at the client end? I'm also going to tell Asterisk to bind to all IPs and then restart it when there's no calls in progress, perhaps that's all I need to do? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.
On 2017-04-18 03:39 PM, Sebastian Nielsen wrote:
You need to ensure that traffic to the SIP box is sent to the
correct IP. Also if you use split-tunnel (eg: not redirect-gateway
def1) you must make sure NAT and traffic redirection works as is
so the Asus router knows it should send the traffic through tunnel
and not via WAN.
I'm not that well versed in OpenVPN, but it's worth noting that we
have the `push "redirect-gateway def1 bypass-dhcp"` directive set on
the server. I have two independent DHCP servers on either side of
the VPN, so that the clients are getting their IP addresses for
their appropriate networks - 192.168.0.0/24 on the server side, and
192.168.1.0/24 on the client side.
IMPORTANT: Then you must, in the ASUS RT-N66U make a port
forward inwards from TUN to the phone client.
I'll give that a shot, but it will have to wait until tomorrow. :)
I would suggest wiresharking on the client side and see which
IP Asterisk suggest the client should connect back to. This
should be the internal IP of the asterisk server as seen from
the openvpn server's point of view.
Another important thing: The local network in the Openvpns
machine locatiin, may NOT have same subnet as the network behind
the asus.
All these must be separate, like:
server network: 192.168.1.0/24
Openvpn tunnel network: 192.168.2.0/24
Asus network: 192.168.3.0/24
I'm pretty sure that I've got this subnet separation in place. If I
didn't cover it in my original post, the network looks like this:
Server network: 192.168.0.0/24
OpenVPN network: 10.8.0.0/24
Asus network: 192.168.1.0/24
The Asterisk SIP registration appears to be responding properly to
this - this is what I see when I do a 'sip show peer' for an Aastra
phone that's connecting through the VPN (Asterisk output is
truncated):
ToHost :
Addr->IP : 10.8.0.6:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: FrontDesk1
SIP Options : (none)
Codecs : (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing : No
Status : Unmonitored
Useragent : Aastra 6731i/3.2.2.1136
Reg. Contact : sip:FrontDesk1@10.8.0.6:5060;transport=udp
Else you get bizarre routing problems when states appear in
the state table.
Originalmeddelande ----
Från: Ernie Dunbar <maill...@lightspeed.ca>
Datum: 2017-04-19 00:25 (GMT+01:00)
Till: 'Asterisk Users Mailing List - Non-Commercial
Discussion' <asterisk-users@lists.digium.com>
Rubrik: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice.
Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.
First, here's our technical details:
The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a
NAT router. The router has UDP port 1194 forwarded to our server.
This server also runs our office Asterisk PBX, so there isn't any
networking hardware or firewall between the VPN tunnel and the
Asterisk PBX.
The OpenVPN client is an Asus RT-N66U router, which if I'm not
mistaken, runs a somewhat modified version of Tomato.
I've got the VPN tunnel working well enough. I can do practically
anything from a computer hooked up to the client router as if I
were in the main office where the server is. But any SIP client I
use - whether it's a hardware SIP phone or a soft phone like
Zoiper, can connect to the Asterisk server without issue. Making
calls can work, accepting calls works, but I only get 1 way voice
traffic. I can hear voice data coming in FROM the Asterisk PBX,
but I cannot send any.
In my experience with SIP, this usually means a firewall is
breaking the connection from the client phone to the Asterisk
server. I just can't for the life of me find what could be wrong.
None of the other traffic is being blocked. The ipfw firewall on
the Asterisk PBX is extremely open (see below). The firewall on
the client router is turned off, and as far as I can tell, most
NAT routers don't even block outbound traffic in the first place.
I can't see how traffic from the TUN interface on the OpenVPN
server even can be blocked going to another IP address on the same
box, but here are the IPFW rules:
root@ldinfo:/etc/asterisk# iptables -L -n
Chain INPUT (policy ACCEPT)
[asterisk-users] SIP connections over OpenVPN connection get one-way voice.
Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't working *quite* as well as we'd hoped. First, here's our technical details: The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT router. The router has UDP port 1194 forwarded to our server. This server also runs our office Asterisk PBX, so there isn't any networking hardware or firewall between the VPN tunnel and the Asterisk PBX. The OpenVPN client is an Asus RT-N66U router, which if I'm not mistaken, runs a somewhat modified version of Tomato. I've got the VPN tunnel working well enough. I can do practically anything from a computer hooked up to the client router as if I were in the main office where the server is. But any SIP client I use - whether it's a hardware SIP phone or a soft phone like Zoiper, can connect to the Asterisk server without issue. Making calls can work, accepting calls works, but I only get 1 way voice traffic. I can hear voice data coming in FROM the Asterisk PBX, but I cannot send any. In my experience with SIP, this usually means a firewall is breaking the connection from the client phone to the Asterisk server. I just can't for the life of me find what could be wrong. None of the other traffic is being blocked. The ipfw firewall on the Asterisk PBX is extremely open (see below). The firewall on the client router is turned off, and as far as I can tell, most NAT routers don't even block outbound traffic in the first place. I can't see how traffic from the TUN interface on the OpenVPN server even can be blocked going to another IP address on the same box, but here are the IPFW rules: root@ldinfo:/etc/asterisk# iptables -L -n Chain INPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- 192.168.0.0/24 192.168.0.3 ACCEPT all -- 192.168.1.0/24 192.168.0.3 ACCEPT all -- 10.8.0.0/24 192.168.0.3 ACCEPT all -- X.X.X.X 192.168.0.3 ACCEPT all -- 192.168.0.3 X.X.X.X ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194 REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with icmp-port-unreachable Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Chain POSTROUTING (0 references) target prot opt source destination 192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server are on. 192.168.1.0/24 is the network that the remote router is on. 10.8.0.0/24 is the network that the TUN device creates. X.X.X.X is our datacenter. 192.168.0.3 is the IP address of our PBX. Any assistance would be greatly appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC freezing Asterisk 13
On 2016-07-14 16:40, Joshua Colp wrote: Saint Michael wrote: Many people are reporting the same issue, so it is not my imagination. Asterisk 13 above 13.1 is useless for anybody who relies on res_odbc.so. As you know, after that version, the dropped the complexity of Pooling onto unix_odbc itself. Not so simple, it seems. I noticed that after a few hours of inactivity, any call to func_odbc-defined funcions will block and hang for ever. All we can do at that point is reset Asterisk. I think it was highly rushed a decision to drop all the work done in ODBC inside Asterisk. Maybe unix_odbc pooling is not ready, has bugs, it cannot be used in production. I don't know what the issue is, but I had to downgrade to Asterisk 13.1 and my ODBC problems disappeared. Asterisk did not need to drop the ODBC pooling code. It did work. It should be fixed, made faster, etc. This has already been done[1] and will be released in Asterisk 13.10, which just had an rc3 released. I also sent an email to the list[2] when the fix went in. These fixes have continued to show no problems themselves although they just exposed an issue with func_odbc which was fixed in the rc3 that was just released. There's no issues open currently against that work. As for the res_odbc changes themselves which exposed problems in UnixODBC those went in as of Asterisk 13.8[3], not earlier. Prior to 13.8 there was no pooling at all. [1] http://blogs.asterisk.org/2016/06/15/asterisk-odbc-connections/ [2] http://lists.digium.com/pipermail/asterisk-users/2016-June/289326.html [3] http://blogs.asterisk.org/2016/02/17/odbc_gutting/ Jumping Jesus on a pogo stick. And here, I was trying to build a version of Asterisk 13.8-cert that 1) absolutely required ODBC because I couldn't even build res_config_mysql.so anymore, and 2) I was trying to get ODBC working on Ubuntu 16.04, in spite of it being, um, apparently completely missing from the OS. Well, I guess I can give up *that* fool's quest! This post has been most illuminating! -- _ -- 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] Compile of smsq.c failed on Ubuntu Xenial (16.04LTS)
On 2016-07-13 17:09, Ernie Dunbar wrote: Hi everyone. I'm trying to compile Asterisk with the smsq utility on Ubuntu 16.04 LTS, and while most things are compiling fine, smsq fails with the following output: root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq [CC] smsq.c -> smsq.o [LD] smsq.o strcompat.o -> smsq strcompat.o: In function `_ast_malloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:535: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:535: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `_ast_realloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' strcompat.o: In function `_ast_strdup': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:624: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:624: undefined reference to `ast_log' strcompat.o: In function `_ast_strndup': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:654: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:654: undefined reference to `ast_log' strcompat.o: In function `_ast_vasprintf': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:694: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:694: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `_ast_realloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `ast_str_set_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1014: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_append_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1032: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_set_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1014: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_append_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1032: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_set_substr': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1039: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_append_substr': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1046: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_set_escapecommas': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1053: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_append_escapecommas': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1060: undefined reference to `__ast_str_helper2' collect2: error: ld returned 1 exit status ../Makefile.rules:163: recipe for target 'smsq' failed make: *** [smsq] Error 1 Years and years of installing binary packages have made my make-fu weak, but I've surmised that it's having trouble loading the asterisk.h library. To get this far, I modified smsq.h to specify the path of asterisk.h to say: #include "../include/asterisk.h" But now I get the output we see above. Perhaps there's an easier way to make it find the include files it needs? Through trial and error, I've fo
[asterisk-users] Compile of smsq.c failed on Ubuntu Xenial (16.04LTS)
Hi everyone. I'm trying to compile Asterisk with the smsq utility on Ubuntu 16.04 LTS, and while most things are compiling fine, smsq fails with the following output: root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq [CC] smsq.c -> smsq.o [LD] smsq.o strcompat.o -> smsq strcompat.o: In function `_ast_malloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:535: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:535: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `_ast_realloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' strcompat.o: In function `_ast_strdup': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:624: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:624: undefined reference to `ast_log' strcompat.o: In function `_ast_strndup': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:654: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:654: undefined reference to `ast_log' strcompat.o: In function `_ast_vasprintf': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:694: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:694: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `_ast_realloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `ast_str_set_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1014: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_append_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1032: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_set_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1014: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_append_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1032: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_set_substr': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1039: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_append_substr': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1046: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_set_escapecommas': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1053: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_append_escapecommas': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1060: undefined reference to `__ast_str_helper2' collect2: error: ld returned 1 exit status ../Makefile.rules:163: recipe for target 'smsq' failed make: *** [smsq] Error 1 Years and years of installing binary packages have made my make-fu weak, but I've surmised that it's having trouble loading the asterisk.h library. To get this far, I modified smsq.h to specify the path of asterisk.h to say: #include "../include/asterisk.h" But now I get the output we see above. Perhaps there's an easier way to make it find the include files it needs? --
[asterisk-users] Trying to record incoming calls that go to queues in Asterisk v11
Hi everyone.
It seems that all the documentation for Asterisk has become obsolete
when it comes to using the Monitor command on a call queue.
To the best of my knowledge, the way to get Asterisk to record a call
that goes into one of your call queues is by doing this in the dialplan:
exten => 1,1,Answer()
same => n,Set(DAY=${STRFTIME(${EPOCH},,%Y-%m-%d)})
same => n,Set(TIME=${STRFTIME(${EPOCH},,%H%M%S)})
same => n,Set(MONITOR_FILENAME="incoming/${DAY}/${TIME}-${EXTEN}")
same => n,Queue(lsqueue-all)
same => n,Macro(handle-hangup)
and then add these lines in the queue context in queues.conf:
[lsqueue-all]
monitor-format=wav
monitor-type=MixMonitor
But when I reload Asterisk, I don't get any new files in
/var/spool/asterisk/monitor/incoming.
We're able to record our outgoing calls without any trouble with the
following dialplan:
exten => call,1,NoOp()
same => n,Set(DAY=${STRFTIME(${EPOCH},,%Y-%m-%d)})
same => n,Set(TIME=${STRFTIME(${EPOCH},,%H%M%S)})
same => n,Set(FILENAME="outgoing/${DAY}/${TIME}-${E}")
same => n,Monitor(wav,${FILENAME},m)
same => n,Dial(SIP/dolphintel/${E})
The file permissions on the "outgoing" and "incoming" are the same, plus
we don't get any errors in the Asterisk console about not being able to
write the files, so I'm pretty sure it's not a problem with actually
writing the files. It just doesn't seem to even try.
Any help will be much appreciated.
--
_
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Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.
On 2016-02-17 16:28, Richard Mudgett wrote: On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar <maill...@lightspeed.ca> wrote: On 2016-02-17 15:32, Richard Mudgett wrote: On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar <maill...@lightspeed.ca> wrote: Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old version has been working, and was compiled on this same machine with this same operating system. This is the error I get when doing the make: [CC] res_fax_spandsp.c -> res_fax_spandsp.o res_fax_spandsp.c: In function ‘spandsp_v21_new’: res_fax_spandsp.c:487:52: error: ‘MODEM_CONNECT_TONES_FAX_CED_OR_PREAMBLE’ undeclared (first use in this function) This is defined by spandsp itself in one of its headers. Have you installed the spandsp dev package? Richard Yes, but it's occurred to me that I'm not using the latest version of the Debian spandsp-dev package. However, at the same time, I'm not exactly compiling the latest version of Asterisk 11 either. Is it that big of a deal, or should I try to pin the package from the next version of Debian? The change that started using the define was made on Dec 28, 2011 to improve V.21 preamble detection. (Git change fdda4947767a5c0ee2424532ff5f01250797175d ) The spandsp version that compiles on my system is 0.0.6~pre12-1 Maybe you have remnants of an older version of spandsp still installed. Richard Okay, I've gotten it compiled. The problem was that Debian's version of libspandsp-dev wasn't the right one. I downloaded and installed the latest version found on spandsp's website, and the compile went through without any issue. -- _ -- 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] Problem compiling res_fax_spandsp.c on Debian server.
On 2016-02-17 15:32, Richard Mudgett wrote: On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar <maill...@lightspeed.ca> wrote: Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old version has been working, and was compiled on this same machine with this same operating system. This is the error I get when doing the make: [CC] res_fax_spandsp.c -> res_fax_spandsp.o res_fax_spandsp.c: In function ‘spandsp_v21_new’: res_fax_spandsp.c:487:52: error: ‘MODEM_CONNECT_TONES_FAX_CED_OR_PREAMBLE’ undeclared (first use in this function) This is defined by spandsp itself in one of its headers. Have you installed the spandsp dev package? Richard Yes, but it's occurred to me that I'm not using the latest version of the Debian spandsp-dev package. However, at the same time, I'm not exactly compiling the latest version of Asterisk 11 either. Is it that big of a deal, or should I try to pin the package from the next version of Debian? -- _ -- 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
[asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.
Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old version has been working, and was compiled on this same machine with this same operating system. This is the error I get when doing the make: [CC] res_fax_spandsp.c -> res_fax_spandsp.o res_fax_spandsp.c: In function ‘spandsp_v21_new’: res_fax_spandsp.c:487:52: error: ‘MODEM_CONNECT_TONES_FAX_CED_OR_PREAMBLE’ undeclared (first use in this function) res_fax_spandsp.c:487:52: note: each undeclared identifier is reported only once for each function it appears in res_fax_spandsp.c: In function ‘spandsp_fax_gateway_start’: res_fax_spandsp.c:819:36: error: ‘t38_gateway_state_t’ has no member named ‘t38’ res_fax_spandsp.c:870:43: error: ‘t38_gateway_state_t’ has no member named ‘t38’ res_fax_spandsp.c:870:83: error: ‘t38_gateway_state_t’ has no member named ‘t38’ make[1]: *** [res_fax_spandsp.o] Error 1 make: *** [res] Error 2 It sounds like I need a library that doesn't exist on this system, but I can't find anything in the includes for this file that would suggest that something is missing. -- _ -- 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] Which router/firewall would you use for a virtual-PBX Asterisk installation?
Oh, don't worry about us going cheap on security. We use A2Billing (along with some Fail2Ban configuration for bad logins) to limit the number and cost of calls that can go out through a compromised SIP account, so that when, not *if*, a customer's SIP account gets compromised, the attacker gets cut off at the knees before they can even get out the door. We've even added bogus connection charges on international calls that get removed before we bill our customers, to speed up the process and reduce our losses even further. Our customers are even happy that these billing limits are in place. No, this is all about playing nice with our load balancing software and protecting databases and backend servers that have no business being available to the public. But mostly it's about the load balancer (IPTables on said servers can take care of "visible to the public). I just want to make sure that the router we use will play nice with Asterisk, since we've already seen network hardware that looks good on paper, but fails miserably in practice. In fact, we see it so often with individual customers' crap routers causing voice quality issues, that by default we don't trust simple math. So here I am, asking everyone what router they use, and whether you're happy with the results when there's 100 simultaneous SIP calls in progress. I want to know what happens when the rubber hits the road. On 2015-11-20 14:22, Telium Technical Support wrote: Well router and firewall are very different...it depends on what you are trying to accomplish. If you are trying to secure an Asterisk-based call center, get a real security product. Look here for details: http://www.voip-info.org/wiki/view/Asterisk+security This covers firewall, Asterisk lock-down, and Asterisk specific security. The average break-in/fraud cost is $25,000 per day. (watch the Astricon videos for more details). So going cheap on security isn't a smart move for a commercial installation. If you just want a router/switch, figure out the simultaneous call capacity x codec demands in bps, and there is your backplane switching speed requirements. Even with 100 simultaneous calls at g711, a lower end Cisco (3xx) router/switch will have no problem. -M- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar Sent: Friday, November 20, 2015 3:25 PM To: Asterisk Users Subject: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation? Hi everyone. We've got a fairly large base of customers who use our Asterisk server for phone service in a virtual PBX kind of way, where the server is security hardened and exposed to the internet for them to connect to remotely with SIP and IAX. It's certainly not the sort of affair where we're running it as a PBX just within the building. As a result, we see network traffic coming through eth0 between 512 Kbps and about 3.0 Mbps, depending on the time of day. We haven't so far been using a hardware firewall/router on our server network, but it's becoming increasingly clear that we need to. We have enough experience to know that Asterisk is pretty sensitive when it comes to network hardware in our situation - we've had to replace one otherwise perfectly good 100 Mbps network switch because it simply wasn't able to keep up with the amount of streaming audio we put through it, and it badly affected voice quality. We have other traffic flowing through our server network too, including a significant amount of e-mail and web traffic, although that's not quite as sensitive to the quality of our network hardware. If you've got these large requirements for Asterisk, I'd love to hear what you use for a router, and whether that router has met your needs. It would also be nice to hear about what kinds of routers to avoid that you may have tried in the past and found lacking. -- _ -- 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
[asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?
Hi everyone. We've got a fairly large base of customers who use our Asterisk server for phone service in a virtual PBX kind of way, where the server is security hardened and exposed to the internet for them to connect to remotely with SIP and IAX. It's certainly not the sort of affair where we're running it as a PBX just within the building. As a result, we see network traffic coming through eth0 between 512 Kbps and about 3.0 Mbps, depending on the time of day. We haven't so far been using a hardware firewall/router on our server network, but it's becoming increasingly clear that we need to. We have enough experience to know that Asterisk is pretty sensitive when it comes to network hardware in our situation - we've had to replace one otherwise perfectly good 100 Mbps network switch because it simply wasn't able to keep up with the amount of streaming audio we put through it, and it badly affected voice quality. We have other traffic flowing through our server network too, including a significant amount of e-mail and web traffic, although that's not quite as sensitive to the quality of our network hardware. If you've got these large requirements for Asterisk, I'd love to hear what you use for a router, and whether that router has met your needs. It would also be nice to hear about what kinds of routers to avoid that you may have tried in the past and found lacking. -- _ -- 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
[asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?
This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan was broken by me misspelling the correct dialplan to go to. Then our Boxing Day dialplan was broken when I copied and pasted the correct dialplan from one similar extension number to the other, like this: ; Christmas ; exten = 821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early,s,1) exten = 821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1) exten = 821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1) exten = 821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1) then failed to notice the problem until it was too late. Of course, that only applied on Boxing day and couldn't be noticed earlier, either. I suppose the first problem where I misspelt the dialplan can be solved by testing the dialplan in another extension and modifying the time to now + 2 minutes. But how can I avoid stupid errors in the extension number, when testing by definition requires that I change the extension number to and fro? This appears to boil down to always remember to test it at the time that it becomes relevant. But if I was the kind of person who always remembered to do things at the right time, then there would never be a need for computers to do jobs like this in the first place. -- _ -- 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] PRI trunk between Asterisk servers does not work.
Quoting Ioan Indreias indre...@gmail.com: On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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 Hello Ernie, Could you post the dahdi/system.conf from both voip1 and voip3 servers? I suspect that you have not correctly defined the data channel (dchan setup should be in system.conf and not in chan_dahdi.conf, where I see a not necessarily dchannel configuration) HTH, Ioan Okay, here's /etc/dahdi/system.conf (it's unmodified from the autogenerated file): # Autogenerated by /usr/sbin/dahdi_genconf on Mon Jul 26 22:53:04 2010 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,esf,b8zs # termtype: te bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,0,0,esf,b8zs # termtype: te bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,0,0,esf,b8zs # termtype: te bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global data loadzone = us defaultzone = us This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Tony Mountifield t...@softins.co.uk: In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? It looks like this: # asterisk -rx 'pri show span 3' Primary D-channel: 72 Status: Provisioned, Down, Active Switchtype: National ISDN Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Logical Channel Mapping: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 The only differences I see between 'pri show span 3' and 'pri show span 4' are that the status on span 4 is Provisioned, Up, Active and that the D-channel is different, which is to be expected. It might be useful to see the complete chan_dahdi.conf from Voip1. To save space, you can list it without comments like this: # grep -v '^;' /etc/asterisk/chan_dahdi.conf Okay, here you go: [channels] usecallerid=yes cidsignalling=bell cidstart=polarity facilityenable=yes hidecallerid=no callwaitingcallerid=yes callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=no immediate=no group=1 signalling=pri_cpe switchtype=national pridialplan=unknown relaxdtmf=yes context=local channel=1-23 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=3600 #include dahdi-channels.conf And dahdi-channels.conf looks like: group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group=4 context=default switchtype = national signalling = pri_net channel = 73-95
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Tony Mountifield t...@softins.co.uk: In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] PRI trunk between Asterisk servers does not work.
Quoting Tim Nelson tnel...@rockbochs.com: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) Excellent! Funny thing about that. Our original plan was to use a SIP trunk until we discovered that faxes don't work worth a damn that way. Ergo, I didn't compile libpri first. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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
[asterisk-users] PRI trunk between Asterisk servers does not work.
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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
[asterisk-users] Difference between Asterisk/libPRI/DAHDI versions breaks Caller ID?
Hi List! We have two Asterisk servers connected to a PRI, an old one and a new one. The old server (voip1 let's call it) is running Asterisk 1.4.23, libpri 1.4.9, and DAHDI 2.1.0.4. The new server (voip2) is running Asterisk 1.8.6, libpri 1.4.2 and DAHDI 2.4.0. We've had serious, show-stopping stability issues with any previous version of Asterisk 1.8, and we're very leery about changing to any other version now that we've found one that works. Voip1 used to be our live SIP hosting server, and while we were migrating to the new Asterisk server, the old server continued to accept calls on the PRI and forwarded those calls to the new server to minimize disruption to service. Since our intention was to move the new server to the PRI when the migration was fully completed, we set up Voip1 as a PRI bridge - one port on the PRI card was used for connecting directly to the PRI, and another port was set up as an exact copy of our PSTN provider's PRI (our configuration below will show this), and we connected this from Voip1 to Voip2. Incoming DIDs would have a dialplan on Voip1 that accepted the call and then immediately called the same DID on Voip2, like this: exten = 6041234567,1,Dial(DAHDI/g4/6041234567) exten = 6041234567,n,Macro(handle-hangup,hc-16) This worked flawlessly. All of our DIDs were migrated to Voip2 and everything was ironed out over time. The problem is, that apparently the PRI between Voip1 and Voip2 is not exactly the same as the PRI from our PSTN provider. I should have been able to simply unplug Voip1 and plug the PRI directly into Voip2 without any issues whatsoever, but instead we found the following: 1. Incoming calls worked fine for two-way voice, but any call coming through the PRI would show Anonymous for the caller ID name and number. This was in spite of the fact that first, this did not happen when the PRI went through Voip1, and second, after setting up the dialplan on our inbound context to display the Caller ID name and number on the Asterisk console, I could see the name and number as it was supposed to be displayed. 2. After googling for this issue, I found this page: http://forums.asterisk.org/viewtopic.php?p=166952 which described a solution to change sip.conf to enable trustrpid and sendrpid globally. This fixed the problem somewhat, but many of our customers ceased to be able to make outbound calls because their ATAs were now sending either a MAC address as their caller ID name and number, or their SIP usernames as the caller ID name and number. Something somewhere wouldn't allow outbound calling as a result, but I think the main problem is that we would trust and/or send the RPID not just to the SIP phones, but also *from* the SIP phones. We should be able to set the caller ID name and number on the server, rather than trust the client. Either way, these settings weren't interfering when the equipment on the other side of the PRI was an Asterisk box, so I don't understand why Asterisk's behaviour had changed at all. Okay, finally, I have the DAHDI configuration for both servers, to show how the PRI was set up: Voip1: [channels] usecallerid=yes cidsignalling=bell ; This one causes problems with Bell telephone numbers, or something. cidstart=polarity ; The default is ring ; cidstart=ring facilityenable=yes hidecallerid=no callwaitingcallerid=yes callwaiting=no threewaycalling=yes transfer=yes echocancel=yes ;echotraining=yes echocancelwhenbridged=no immediate=no ; PRI group 1: group=1 signalling=pri_cpe switchtype=national pridialplan=unknown relaxdtmf=yes context=local channel=1-23 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=3600 #include dahdi-channels.conf FILE dahdi-channels.conf: ; Autogenerated by /usr/sbin/dahdi_genconf on Mon Jul 26 22:53:04 2010 -- do not hand edit ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings ; ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) ;group=0,11 ;context=from-pstn ;switchtype = national ;signalling = pri_cpe ;channel = 1-23 ;context = default ;group = 63 ; Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 ;group=0,12 ;context=from-pstn ;switchtype = national ;signalling = pri_cpe ;channel = 25-47 ;context = default ;group = 63 ;; Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 ;group=0,13 ;context=from-pstn ;switchtype = national ;signalling = pri_cpe ;channel = 49-71 ;context = default ;group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 FILE /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Mon Jul 26 22:53:04 2010 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,esf,b8zs # termtype:
[asterisk-users] No audio format found to offer.
This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error: -- AGI Script Executing Application: (DIAL) Options: (SIP/t564/1XX4332,,HR) == Using SIP RTP CoS mark 5 [Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio format found to offer. Cancelling call to 1XX4332 -- Couldn't call t564/1XX332 == Everyone is busy/congested at this time (0:0/0/0) I've checked to ensure that both formats are loaded into Asterisk: voip2*CLI module show like 729 Module Description Use Count format_g729.so Raw G729 data0 1 modules loaded voip2*CLI module show like 723 Module Description Use Count format_g723.so G.723.1 Simple Timestamp File Format 0 1 modules loaded So I'm at a bit of a loss as to why Asterisk is complaining that there's no audio format found to offer. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] No audio format found to offer.
Quoting Carlos Chavez cur...@telecomabmex.com: The disallow line must be set before any allow line. Since Asterisk has no official G723 support you should not even be trying to use that. That's fantastic. I'll tell that to our SIP trunk provider right away. Do you have the G.279 codec and license installed in your system? Remember that G.729 is not included in Asterisk (as a codec) so it only works in passthru. So G.729 will only work for this trunk if the customer's ATA is using it too? You need to purchase some licenses and install the codec for it to work. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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
[asterisk-users] Outgoing calls get dropped on high-latency connections.
We're a VoIP provider essentially competing with our local incumbent Telco, and a sizeable number of our customers use satellite internet. As a result, these customers never have ping times less than 500ms, and are often exceeding 2500ms. I manually apply a patch to the Asterisk source code every time we upgrade Asterisk, described here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html This change allows our satellite customers to maintain their SIP connection for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and this version seems to have one very strange bug on these high latency connections. On outgoing and *only* outgoing calls, the call drops after two or three minutes. Incoming calls do not have this problem, so I don't think it's the SIP connection getting killed due to a slow INVITE response. Has anyone heard of this bug? Or should I submit a new bug report to the Asterisk project? This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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
[asterisk-users] Using Dial() on SIP and DAHDI connections simultaneously
I think there is a bug in the Dial() application in Asterisk 1.6.2.17 that wasn't present in 1.4.23.1, and I'd like to see if anyone else has this problem. I've been able to reproduce this error: When you use the Dial() command to send a call to both a SIP connection and a DAHDI connection, if the DAHDI connection is busy, the call always gets rejected with an Asterisk message saying the line is busy, even if the SIP connection is not busy. The inverse is not true: if the SIP connection is busy but the DAHDI connection is not, the call goes through to the DAHDI connection without a problem. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] Outgoing calls get dropped on high-latency connections.
Yes, these are our session-timer settings in sip.conf: session-timers=originate session-expires=600 session-minse=90 session-refresher=uas Quoting Faisal Hanif fai...@vopium.com: Have you tried SIP session timer values in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar Sent: Tuesday, June 28, 2011 9:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing calls get dropped on high-latency connections. We're a VoIP provider essentially competing with our local incumbent Telco, and a sizeable number of our customers use satellite internet. As a result, these customers never have ping times less than 500ms, and are often exceeding 2500ms. I manually apply a patch to the Asterisk source code every time we upgrade Asterisk, described here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html This change allows our satellite customers to maintain their SIP connection for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and this version seems to have one very strange bug on these high latency connections. On outgoing and *only* outgoing calls, the call drops after two or three minutes. Incoming calls do not have this problem, so I don't think it's the SIP connection getting killed due to a slow INVITE response. Has anyone heard of this bug? Or should I submit a new bug report to the Asterisk project? This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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 This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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
[asterisk-users] ATA refuses to answer a call?
I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the internet without any firewalling. - When people call this extension, the console shows that Asterisk accepts the call from the DAHDI channel, executes the SIP call, then... nothing. It either waits until the timeout set in the dialplan is up, then goes to voicemail (next step), or it sends a 'hangup cause 102' to the DAHDI channel. Conspicuously missing is the console saying SIP/username is ringing. The following is redacted output from such a call: -- Executing [6045551212@local:1] Dial(DAHDI/6-1, SIP/sipuser|20) in new stack -- Called sipuser -- Accepting call from '7785550001' to '6045551212' on channel 0/6, span 1 -- Channel 0/6, span 1 got hangup, cause 102 == Spawn extension (local, 6045551212, 1) exited non-zero on 'DAHDI/6-1' -- Hungup 'DAHDI/6-1' -- No one is available to answer at this time (1:0/0/0) -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
On 29/04/11 5:06 AM, Ira wrote: At 05:56 AM 4/28/2011, you wrote: If I can install 1.8 and know that I can turn off things to get to 1.4 solidness, then I don't have a problem with this kettle of fish. BTW, where does 1.10 fit into this conversation? Personally, 1.8 has never lasted more than 12 hours on my box without dying and once I figured out how it dies, every beta and every release will fail within moments if I followed the same very short test script. I did put up a bug report on the problem once and was told within moments it wasn't a bug, but I'm not smart enough to understand what I'm supposed to do to troubleshoot and the same configuration has always run on 1.2, 1.6 and 1.10 so from my perspective, it's a bug. What's the URL to the bug you submitted? I'm running 1.8 here 24/7 with no problems other than the ones that Alec Davis fixed. I've got it running in I think 4 installations and we're not getting any core dumping or anything - obviously I'm only using a subset of the full functionality and most modules are not included. What features do you have disabled? It would be helpful to know this for future 1.8 implementation, although right now we can't quite use it yet. -- _ -- 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] Forwarding XXXX to XXXX prevented.
So... no solution to this problem? It does depend on how you set up the call forwarding on asterisk and sometimes when the ATA sends the forwarding call to the Voip provider server it has nothing to do with it which causes a problem. if you disable call forwarding remotely see if that works also. its a tricky situation. On Wed, 2011-03-23 at 16:18 -0700, Ernie Dunbar wrote: I have a Linksys 2102 ATA here that does call forwarding internally with the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the call properly. This is what shows up in the console when an incoming call is made while the ATA is call-forwarded: -- Called Username -- Got SIP response 302 Moved Temporarily back from XX.XXX.XX.XXX -- Now forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' (thanks to SIP/Username-0045) -- Forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' prevented. == Everyone is busy/congested at this time (1:1/0/0) The SIP configuration allows call forwarding (cancallforward=yes), so I'm at a loss as to what is preventing the forwarding. It's not like Asterisk is very specific about that. -- _ -- 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
[asterisk-users] Forwarding XXXX to XXXX prevented.
I have a Linksys 2102 ATA here that does call forwarding internally with the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the call properly. This is what shows up in the console when an incoming call is made while the ATA is call-forwarded: -- Called Username -- Got SIP response 302 Moved Temporarily back from XX.XXX.XX.XXX -- Now forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' (thanks to SIP/Username-0045) -- Forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' prevented. == Everyone is busy/congested at this time (1:1/0/0) The SIP configuration allows call forwarding (cancallforward=yes), so I'm at a loss as to what is preventing the forwarding. It's not like Asterisk is very specific about that. -- _ -- 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
[asterisk-users] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers = mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI realtime mysql status localhost configured for mya2billing@localhost, port 3306 with username a2billinguser. mya2billing configured for mya2billing@localhost, port 3306 with username a2billinguser. [Mar 7 14:38:59] ERROR[15943]: res_config_mysql.c:1575 mysql_reconnect: MySQL RealTime: Failed to connect database server mya2billing on localhost (err 2002). Check debug for more info. [Mar 7 14:38:59] ERROR[15943]: res_config_mysql.c:1575 mysql_reconnect: MySQL RealTime: Failed to connect database server mya2billing on localhost (err 2002). Check debug for more info. This doesn't make any sense. res_mysql.conf contains working mysql credentials that I can verify with running mysql from the command line. -- _ -- 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 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config stanza and see if that helps (or whatever is the actual location of your mysql.sock file). Hmm. This appears to have fixed the problem, even though I swear I've done this already. (and for anyone reading this, on Debian the file is mysqld.sock) -- _ -- 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
[asterisk-users] PRI B-Channel restarting itself continually
On our live server, running Asterisk 1.4.23.1 and DAHDI-Linux 2.1.0.4. On occasion (not too rare, happens maybe once every month or two), the PRI and/or DAHDI will stop working properly and we'll get repeated messages like this: [Feb 25 05:17:22] VERBOSE[9511] logger.c: -- B-channel 0/1 restarted on span 1 [Feb 25 05:17:25] VERBOSE[9511] logger.c: -- B-channel 0/1 restarted on span 1 [Feb 25 05:17:27] VERBOSE[9511] logger.c: -- B-channel 0/1 restarted on span 1 [Feb 25 05:17:30] VERBOSE[9511] logger.c: -- B-channel 0/1 restarted on span 1 What's going on here? Is there a way to fix this or work around the issue? Every unused B-channel already gets restarted every hour automatically by Asterisk. -- _ -- 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] Aastra phones cannot transfer calls?
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote: At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had no problem at all with transfers. Have you considered trying a newer version? Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. It may be a setting on the phone or a SIP setting. I'll investigate this elsewhere but report back about the solution. I also tried this with a 6757i and a 6753i with no problems (blind and attended) on Asterisk 1.6.2.16.1. Have you updated the handset firmware to 2.6.0.2010? What SIP settings do you have in Asterisk? -- _ -- 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] Aastra phones cannot transfer calls?
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote: At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had no problem at all with transfers. Have you considered trying a newer version? Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. It may be a setting on the phone or a SIP setting. I'll investigate this elsewhere but report back about the solution. I also tried this with a 6757i and a 6753i with no problems (blind and attended) on Asterisk 1.6.2.16.1. Have you updated the handset firmware to 2.6.0.2010? What SIP settings do you have in Asterisk? Actually, I found the problem. Allowtransfer is a new SIP setting (it certainly isn't in sip.conf on the old server) and by default it's set to no globally. Changing this to yes fixes the problem. -- _ -- 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
[asterisk-users] Aastra phones cannot transfer calls?
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside extensions on the PSTN. The procedure I use is to accept a call on one phone, press the transfer button, dial the other extension, and then press transfer again after the second extension answers. The transfer appears to work until I press transfer the second time, at which time the Aastra phone gives the error message Transfer Failed. These same phones have no trouble performing this operation on Asterisk v1.4.23.1. Is this an issue with Aastra phones, or is it a problem with Asterisk 1.6? -- _ -- 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] Aastra phones cannot transfer calls?
At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had no problem at all with transfers. Have you considered trying a newer version? I don't know how that got truncated, but we're running v1.6.2.15 -- _ -- 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] Aastra phones cannot transfer calls?
At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had no problem at all with transfers. Have you considered trying a newer version? Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. It may be a setting on the phone or a SIP setting. I'll investigate this elsewhere but report back about the solution. -- _ -- 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
[asterisk-users] Defining what an extension should do after the Dial() command returns busy.
We have a customer who wants to forward an extension to their cell phone, if and only if that extension is unavailable, or when the Dial() command times out. However, should the Dial() command return busy it should go to voicemail instead. As far as I know, the dialplan doesn't support this. Certainly not natively or in any particularly easy or obvious way, and I can't find anything on voip-info.org to suggest that there is. -- _ -- 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] Defining what an extension should do after the Dial() command returns busy.
It's nice to know that you've tried this and are presenting me with a proven solution. FYI, this doesn't work. Neither do any of the following variations: exten = 27,1,Dial(SCCP/foo,10) exten = 27,n-BUSY,Voicemail(27) exten = 27,n-NOANSWER,Dial(DAHDI/g1/5551234) exten = 27,n,Hangup() or exten = 27,1,Goto(ernie,s,1) [ernie] exten = s,1,Dial(SCCP/lightspeed7,10) exten = s,n-BUSY,Voicemail(27) exten = s,n-NOANSWER,Dial(DAHDI/g1/7788391675) exten = s,n,Hangup() or exten = 27,1,Goto(ernie,s,1) [ernie] exten = s,1,Dial(SCCP/lightspeed7,10) exten = s-BUSY,Voicemail(27) exten = s-NOANSWER,Dial(DAHDI/g1/7788391675) exten = s,n,Hangup() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar Sent: Wednesday, February 09, 2011 1:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Defining what an extension should do after the Dial() command returns busy. We have a customer who wants to forward an extension to their cell phone, if and only if that extension is unavailable, or when the Dial() command times out. However, should the Dial() command return busy it should go to voicemail instead. As far as I know, the dialplan doesn't support this. Certainly not natively or in any particularly easy or obvious way, and I can't find anything on voip-info.org to suggest that there is. Perhaps your googling skills need some management - look for S-BUSY, S-NOANSWER. Here's a snippet that might do what they want - exten = s,1,Dial(DAHDI/1/5551212,30) - exten = s,n-BUSY,voicemail(blah) - exten = s,n-UNAVAILABLE,Dial(DAHDI/1/5552323,30) - exten = t,1,Dial(DAHDI/1/5552323,30) Cell -- _ -- 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
[asterisk-users] Inbound SIP calls work, just not when making calls between extensions.
This is a problem that is completely stumping me, and my understanding of Asterisk dialplans tells me this should never be a problem. Moreover, this scenario works on Asterisk 1.4 but not 1.6. We have a customer with several Aastra 6731 phones. They want incoming calls from the PSTN to work and they also want to be able to call each other internally on a special non-DID number (like extensions 311, 312, 313, etc). In the dialplan, both the extensions for their DID and their internal extensions use the same Dial() command. The only difference that I can see is that we make changes to the CallerID Name field and do a little dance with SIPAddHeader() to make the Aastra phones ring differently. This doesn't appear to have any effect on Asterisk, but when the call is made, the phone responds back with SIP response 400 Bad Request. Here's the two dialplans (private details redacted): Internal calls: exten = _312,1,Set(CALLERID(name)=Internal call) exten = _312,n,SIPAddHeader(Alert-Info: info=Bellcore-dr2) exten = _312,n,Dial(SIP/username2,20) exten = _312,n,Voicemail(312,u) exten = _312,n,Macro(handle-hangup) Calls from the PSTN: [Somecompany-IVR-day] exten = s,1,Dial(SIP/username1SIP/username2SIP/username3,20) exten = s,n,Goto(Somecompany-IVR-night,s,1) The errors from Asterisk when internal calls are made: -- Executing [311@somecompany:1] Set(SIP/username3-01b0, CALLERID(name)=Internal call) in new stack -- Executing [311@somecompany2] SIPAddHeader(SIP/username3-01b0, Alert-Info: info=Bellcore-dr2) in new stack -- Executing [311@somecompany3] Dial(SIP/username3-01b0, SIP/username1,20) in new stack == Using SIP RTP CoS mark 5 -- Called username1 -- Got SIP response 400 Bad Request back from XX.XXX.XXX.X -- SIP/username1-01b1 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [311@somecompany4] VoiceMail(SIP/username3-01b0, 311,u) in new stack -- _ -- 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
[asterisk-users] Really wacky problem with internal extensions.
We have an Asterisk server acting as a hosted PBX system for many clients,
and we're going through an upgrade to Asterisk 1.6 by moving our most
important (and complicated) clients one at a time.
But we're having a problem with one customer that I really can't explain.
I can place calls directly to one phone at the customer's location (they
also have an IVR that asks for an extension number), but the customer
cannot do the same internally. All other outbound calls from this
customer, work. The dialplans for the IVR and for internal dialing are
very nearly identical, and making them completely identical doesn't change
anything. The dialplans are pasted at the end of this message.
When the customer dials an internal extension, the Asterisk console
produces this output (usernames redacted):
-- Executing [303@XX:1] Set(SIP/XX2-04ce,
CALLERID(name)=Internal call) in new stack
-- Executing [303@XX:2] GotoIf(SIP/XX2-04ce,
0?dialfw:dial) in new stack
-- Goto (XX,303,8)
-- Executing [303@XX:8] Dial(SIP/XX2-04ce,
SIP/XX3,20,g) in new stack
== Using SIP RTP CoS mark 5
-- Called XX3
-- Got SIP response 400 Bad Request back from 209.53.201.33
-- SIP/XX3-04cf is circuit-busy
Usually this SIP response 400 error is due to the firewall at the
customer's location blocking the incoming connection, but then why would
normal inbound calls work? It's not like the Dial() command for those
inbound calls is any different. This customer hasn't changed any firewall
rules during the changeover, and is forwarding unique ports for each
phone. Furthermore, the SIP configuration for these phones send a
qualification message every 60 seconds to keep any NAT translation alive.
Anyway, here's the dialplan for the IVR (only extensions 302 and 303 are
included for brevity):
[ivr-XX]
exten = s,1,Answer
exten = s,n,Playback(silence/1)
exten = s,n,Background(XX/greeting)
exten = s,n,WaitExten(4)
exten = 302,1,GotoIf(${DB_EXISTS(CFIM/302)}?dialfw:dial)
exten = 302,n(dialfw),Set(extension=${DB(CFIM/302)})
exten = 302,n,Set(wait=${MATH(${DB(NumRing/302)}*6,int)})
exten = 302,n,ExecIf($[${wait} != 0]|Dial|SIP/XX2|${wait}|g|)
exten = 302,n,Dial(DAHDI/g1/${extension},90,g)
exten = 302,n,Macro(handle-hangup)
exten = 302,n(dial),Dial(SIP/XX2,30,g)
exten = 302,n,Voicemail(302,u)
exten = 302,n,Macro(handle-hangup)
exten = 303,1,GotoIf(${DB_EXISTS(CFIM/303)}?dialfw:dial)
exten = 303,n(dialfw),Set(extension=${DB(CFIM/303)})
exten = 303,n,Set(wait=${MATH(${DB(NumRing/303)}*6,int)})
exten = 303,n,ExecIf($[${wait} != 0]|Dial|SIP/XX3|${wait}|g|)
exten = 303,n,Dial(DAHDI/g1/${extension},90,g)
exten = 303,n,Macro(handle-hangup)
exten = 303,n(dial),Dial(SIP/XX3,30,g)
exten = 303,n,Voicemail(303,u)
exten = 303,n,Macro(handle-hangup)
exten = 0,1,Answer
exten = 0,n,SIPAddHeader(Alert-Info: info=Bellcore-dr4)
exten =
0,n,Dial(SIP/XX2SIP/XX3SIP/XX4SIP/XX5SIP/XX6SIP/XX7,25,g)
exten = 0,n,Voicemail(300,u)
exten = 0,n,Macro(handle-hangup)
exten = t,1,Answer
exten = t,n,SIPAddHeader(Alert-Info: info=Bellcore-dr4)
exten =
t,n,Dial(SIP/XX2SIP/XX3SIP/XX4SIP/XX5SIP/XX6SIP/XX7,25,g)
exten = t,n,Voicemail(300,u)
exten = t,n,Macro(handle-hangup)
exten = i,1,Playback(XX/invalid)
exten = i,n,Goto(s,1)
And this is the outgoing dialplan for the customer (for internal lines and
special features)
[XX]
exten = _*98,1,Answer
exten = _*98,n,VoicemailMain()
exten = _*88,1,Answer
exten = _*88,n,VoicemailMain(300)
exten = _*72,1,Answer
exten = _*72,n,Wait(1)
exten = _*72,n,Read(extension,XX/enter-extension,3)
exten = _*72,n,Read(fwdnum,XX/forward-to,10)
exten = _*72,n,Read(numrings,XX/num-of-rings,1)
exten = _*72,n,Set(DB(CFIM/${extension})=${fwdnum})
exten = _*72,n,NoOp(Numrings: ${numrings} ${numrings})
exten = _*72,n,Set(DB(NumRing/${extension})=${numrings})
exten = _*72,n,Playback(XX/your-extension)
exten = _*72,n,SayDigits(${extension})
exten = _*72,n,Playback(XX/will-forward-to)
exten = _*72,n,SayDigits(${fwdnum})
exten = _*72,n,Playback(XX/after)
exten = _*72,n,SayDigits(${numrings})
exten = _*72,n,Playback(XX/rings)
exten = _*72,n,Macro(handle-hangup)
exten = _*73,1,Answer
exten = _*73,n,Wait(1)
exten = _*73,n,Read(extension,XX/enter-extension,3)
exten = _*73,n,Set(${DB_DELETE(CFIM/${extension}))
exten = _*73,n,Playback(XX/cfwd-cancelled)
exten = _*73,n,Macro(handle-hangup)
exten = 302,1,Set(CALLERID(name)=Internal call)
exten = 302,n,GotoIf(${DB_EXISTS(CFIM/302)}?dialfw:dial)
exten = 302,n(dialfw),Set(extension=${DB(CFIM/302)})
exten = 302,n,Set(wait=${MATH(${DB(NumRing/302)}*6,int)})
exten = 302,n,ExecIf($[${wait} != 0]|Dial,SIP/XX2,${wait},g)
exten = 302,n,Dial(DAHDI/g1/${extension},90,g)
exten = 302,n,Macro(handle-hangup)
exten = 302,n(dial),Dial(SIP/XX2,20,g)
exten = 302,n,Voicemail(302,u)
exten = 302,n,Macro(handle-hangup)
exten =
Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog? Not much. In /var/log/asterisk/messages here's a lot of lines like this: [Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from 'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matching peer found And /var/log/syslog has all the normal output from a2billing.php and making calls complete and such. The other funny thing is that except for the massive number of zombie processes, calls are being made and completed just fine. Even voice quality is quite high. -- _ -- 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 1.6 produces *many* zombie processes on Debian.
Actually, no. This is part of a migration, and those are mostly customers' secondary lines (which for the most part, aren't even active). We get a lot of these bad logins because the retry times on the ATAs are quite short. Asterisk really *shouldn't* leave zombies around for every bad login, but if it does, then I suppose cleaning up these missing accounts might fix it. Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Mensagem original - Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog? Not much. In /var/log/asterisk/messages here's a lot of lines like this: [Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from 'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matching peer found And /var/log/syslog has all the normal output from a2billing.php and making calls complete and such. The other funny thing is that except for the massive number of zombie processes, calls are being made and completed just fine. Even voice quality is quite high. -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. Simple In sip.conf please set alwaysauthreject = yes Thanks for the tip, but we already did that a while ago. :) -- _ -- 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
[asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. -- _ -- 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
[asterisk-users] MP3s not decoding properly for MusicOnHold.
I have some MP3 files that play well in any MP3 player I throw at them, but when I try to make a MusicOnHold class with them, I get a continuous stream of errors like this: [Dec 2 13:20:31] WARNING[9120]: mp3/common.c:148 decode_header: Layer 2 not supported! [Dec 2 13:20:31] WARNING[9120]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 50686f74 [Dec 2 13:20:31] WARNING[9120]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame e7becffc [Dec 2 13:20:31] WARNING[9120]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443302 I figured this was something that was answered before, but googling for this error message reveals nothing. -- _ -- 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
[asterisk-users] Sending calls to a particular T1 port.
We have two Asterisk servers. One is a live server supporting our customers, and the other is a backup server that's being upgraded and pressed into service. Both servers have a Digium TE405P T1 card in them, and in order to test the T1 service on the backup server, I've created a T1 crossover cable (as per http://www.voip-info.org/wiki/view/crossover+T1+cable) that goes from port 4 on the live server to port 1 on the backup server. Both TE405P's have been configured, and I get a green light on port 4 on the live server and port 1 on the backup server. The problem I have is trying to route calls through this T1 connection. To the best of my knowledge, this configuration on the live server should work: In /etc/asterisk/chan_dahdi.conf: group=4 context=local switchtype = national signalling = pri_cpe channel = 73-95 context = default group = 63 In /etc/asterisk/extensions.conf: exten = _*88,1,Dial(DAHDI/g4/123456789) However, in the Asterisk console, I get this error on the live server: -- Executing [...@lightspeedout:1] Dial(SCCP/lightspeed7-0062, DAHDI/g4/123456789) in new stack [Nov 12 09:24:41] WARNING[1970]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SCCP/lightspeed7-0062' status is 'CHANUNAVAIL' And no messages at all on the backup server, except this one every 4 seconds: [Nov 12 10:08:04] WARNING[4473]: chan_dahdi.c:4169 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Which Googling reveals to be a fairly mundane and harmless warning message (it happens on real T1's apparently, and isn't related to any kind of outage). Also, I'm not entirely sure how to enable changes to chan_dahdi.conf without restarting asterisk or otherwise killing the 15+ DAHDI channels in progress. Doing this at off-peak times is very slow, as I'm asleep during those times, and thus it can only happen once per day. -- _ -- 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] Sending calls to a particular T1 port.
Oh, this is most excellent. Although it means that my google-fu has failed me. ;) On Fri, Nov 12, 2010 at 1:42 PM, Jonathan Thurman jonat...@thurmantech.comwrote: I didn't read the whole thing, but it looks pretty OK at a glance. http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html I hope that helps, Steve Totaro -- _ -- 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
