[asterisk-users] Best strategy to find and solve voice quality problems
Hi, in my company we use an asterisk installation with around 50 soft- and hardphones of all kind. From time 2 time the users (almost only Softphone users) report some voice qualities... mostly echoes. These problems do not occur on all PCs at the same time and since setup of our PBX almost any PC user has gotten these issues. When I come there to check, everything is fine again... and I can´s see anymore, what could have caused the problem... may it be a high network load, or a high cpu usage or whatever... I activated call recording to hear the quality after such missing-quality reports but every call I listened to showed no issues in the recording so I assume the problem is on the client side. Because it is not always the same user or the same PC I think it cannot be a misbehaviour like wrong headset usage or a problem of a single PC. What is the best strategy to find and solve these kind of problems? Are there any (free would be cool) tools that can monitor the pc-state (concerning at least network and cpu- / process usage) over a long period and display the results in an appropriate way? Is there a way under Windows XP / 7 to ensure Bandwidth for VoIP like QoS (google only showed me such settings for Lync or Windows Server machines...? Is there a way under Windows XP / 7 to ensure CPU-Bandwidth for Applications (like VoIP Clients)? Thanks for any hint, yves -- _ -- 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 Fax detection *11.7
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts: 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels: 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
Digium T.38
Licensed Channels: 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0
so that should be ok.
The corresponding dialplan section starts with
[from-sip]
include = inbound
[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)
in the sip.conf i specified
[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes
...
[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912
but all i can see if i try to send a testfax is
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016,
) in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source
address to 123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
) in new stack
-- Executing [12345678912@from-sip:4]
Progress(SIP/abcde-0016, ) in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016,
5) in new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until
answer for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until
answer for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing
Any hints why thats not working?
Best Regards Jakob
--
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Re: [asterisk-users] Asterisk Fax detection *11.7
It is really more interesting the receiving part. Can you paste here?
Leandro
2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts: 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels: 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
Digium T.38
Licensed Channels: 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0
so that should be ok.
The corresponding dialplan section starts with
[from-sip]
include = inbound
[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)
in the sip.conf i specified
[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes
...
[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912
but all i can see if i try to send a testfax is
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016,
) in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source address
to 123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
) in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016,
) in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5)
in new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing
Any hints why thats not working?
Best Regards Jakob
--
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Re: [asterisk-users] Asterisk not receiving call from VPN address
Hi Duncan, We have sip set debug on and nothing is shown, even though tcpdump/ngrep on the same server does. It's very strange. The output of ip address list is: [root]# ip address list 1: lo: LOOPBACK,UP,LOWER_UP mtu 16436 qdisc noqueue state UNKNOWN link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: eth0: BROADCAST,MULTICAST,UP,LOWER_UP mtu 1500 qdisc mq state UP qlen 1000 link/ether 44:1e:a1:4e:2f:b8 brd ff:ff:ff:ff:ff:ff inet 103.y.y.19/24 brd 103.y.y.255 scope global eth0 inet6 fe80::461e:a1ff:fe4e:2fb8/64 scope link valid_lft forever preferred_lft forever 3: eth1: BROADCAST,MULTICAST mtu 1500 qdisc noop state DOWN qlen 1000 link/ether 44:1e:a1:4e:2f:ba brd ff:ff:ff:ff:ff:ff 4: eth2: BROADCAST,MULTICAST mtu 1500 qdisc noop state DOWN qlen 1000 link/ether 44:1e:a1:4f:30:a4 brd ff:ff:ff:ff:ff:ff 5: eth3: BROADCAST,MULTICAST mtu 1500 qdisc noop state DOWN qlen 1000 link/ether 44:1e:a1:4f:30:a6 brd ff:ff:ff:ff:ff:ff 6: tun0: POINTOPOINT,MULTICAST,NOARP,UP,LOWER_UP mtu 1500 qdisc pfifo_fast state UNKNOWN qlen 100 link/[65534] inet 172.x.x.14 peer 172.x.x.13/32 scope global tun0 The output of netstat -rn is: [root]# netstat -rn Kernel IP routing table Destination Gateway Genmask Flags MSS Window irtt Iface 172.x.x.10 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0 172.x.x.13 0.0.0.0 255.255.255.255 UH0 0 0 tun0 172.x.x.1 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0 172.x.x.18 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0 192.168.234.0 172.x.x.13 255.255.255.0 UG0 0 0 tun0 192.168.235.0 172.x.x.13 255.255.255.0 UG0 0 0 tun0 103.y.y.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 169.z.z.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0 172.21.0.0 172.x.x.13 255.255.0.0 UG0 0 0 tun0 10.0.0.0172.x.x.13 255.0.0.0 UG0 0 0 tun0 0.0.0.0 103.y.y.1 0.0.0.0 UG0 0 0 eth0 On 21 January 2014 17:44, Duncan Turnbull dun...@e-simple.co.nz wrote: Cool That looks like it is arriving at Asterisk - are you sure asterisk is not getting it? If you turn on sip debug in asterisk can you see the SIP packets? It maybe asterisk is ignoring them or replying to them but its going out an interface you hadn’t thought of, I have had that a few times. I should have mentioned to print out your route table and ifconfig. Asterisk can reply on a different address to the original destination especially if it came through a tunnel. Often it will be the tunnel interface address. Usually then we set the secondary address as the outbound proxy on the phone so the phone will also respond to it. Cheers Duncan On 21/01/2014, at 7:18 pm, David Cunningham dcunning...@voisonics.com wrote: Hi Duncan, Thank you for your reply. Here's the netstat: [root]# netstat -udpln | grep asterisk udp0 0 0.0.0.0:50000.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 6672/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 6672/asterisk Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Kamailio server: 17:13:17.103771 IP 103.x.x.x.5060 172.y.y.y.5060: SIP, length: 1228 E...@./g.v.INVITE sip:*1@172.y.y.y:5060;transport=udpSIP/2.0 Record-Route: sip:103.x.x.x;lr=on Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0 Via: SIP/2.0/UDP 192.168.1.40:5060 ;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850 From: sip:9067273@103.x.x.x;tag=1880695235 To: sip:*1@103.x.x.x Call-ID: 1898224288 Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Asterisk server: 17:13:17.093676 IP 103.x.x.x.5060 172.y.y.y.5060: SIP, length: 1228 E...?.?/g.v.INVITE sip:*1@172.y.y.y:5060;transport=udpSIP/2.0 Record-Route: sip:103.x.x.x;lr=on Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0 Via: SIP/2.0/UDP 192.168.1.40:5060 ;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850 From: sip:9067273@103.x.x.x;tag=1880695235 To: sip:*1@103.x.x.x Call-ID: 1898224288 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642
Re: [asterisk-users] Read factory0x7f32f4005940 was pretty quick last time, waiting for them
Hi, My problem has been solved.In my case because of CPU High load . On Mon, Jan 20, 2014 at 3:34 PM, mahdieh saeed mahdieh.sa...@gmail.comwrote: Hi every body our Calls are begging dropped for no reason and it starts with the sound quality dropping and then the caller unable to hear our call center agents. Then the call drops or the caller hangs up unable to hear. I could see following lines inside full log -- [Jan 20 15:21:35] DEBUG[14982] audiohook.c: Write factory 0x7f32f4005940 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14988] audiohook.c: Write factory 0x7f32dc00a0d0 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[15013] audiohook.c: Write factory 0x7f32f0016660 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14988] audiohook.c: Write factory 0x7f32dc00a0d0 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14998] audiohook.c: Write factory 0x7f32ec0069e0 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14982] audiohook.c: Write factory 0x7f32f4005940 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14998] audiohook.c: Write factory 0x7f32ec0069e0 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[15013] audiohook.c: Write factory 0x7f32f0016660 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14982] audiohook.c: Write factory 0x7f32f4005940 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14988] audiohook.c: Write factory 0x7f32dc00a0d0 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[15013] audiohook.c: Write factory 0x7f32f0016660 was pretty quick last time, waiting for them. - mixmonitor app is enable for trying to record channels. we are using asterisk 1.8.20 could you help me what is the problem? Best Regards, Mahdieh Saeed -- _ -- 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 not receiving call from VPN address
Hi Larry, No, they are on separate machines. On 21 January 2014 17:54, Larry Moore lmo...@omninet.net.au wrote: Is Kamalio running on the same system as Asterisk? On 21/01/2014 2:41 PM, David Cunningham wrote: Hi Larry, Thanks for the reply. We have all of those settings left out of our sip.conf, so this should allow everything, right? On 21 January 2014 17:38, Larry Moore lmo...@omninet.net.au mailto:lmo...@omninet.net.au wrote: Have you checked your localnet=, deny=, permit=, contactdeny= contactpermit= settings? My 2c worth. On 20/01/2014 10:51 AM, David Cunningham wrote: Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172 tel:15.599557%20172.x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;__transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.__f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;__branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x;tag=__198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 tel:%2B1%20213%20221%201092 UK: +44 (0) 20 3298 1642 tel:%2B44%20%280%29%2020%203298%201642 Australia: +61 (0) 2 8063 9019 tel:%2B61%20%280%29%202%208063%209019 -- _ -- 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 http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Dialing a SIP URI with an ;ext= parameter
I am going to try a Lync server/asterisk integration, so I really appreciate! Leandro 2014/1/21 Lincoln King-Cliby linc...@controlworks.com Ok, so now I just feel kind of stupid. After I got home I decided to play with this a little more. After far too long I realized that part of the issue was Asterisk parsing the ; as a beginning of a comment (hindsight=duh). A little bit more experimenting and (though I could swear I tried this before) replacing the ; with \; works. That is, to dial a E.164 normalized number with an extension configured as tel:+14404491100;ext=1407 +14404491100;ext=1407 with the SIP Peer for the Lync mediation server named “lync” the working dial() is Dial(SIP/lync/+14404491100\;ext=1407) Hope this may save someone else time down the road. -- Lincoln King-Cliby, CTS, DMC-D Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Silver) V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com Crestron Services Provider *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lincoln King-Cliby *Sent:* Monday, January 20, 2014 5:04 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dialing a SIP URI with an ;ext= parameter Hi All, In the midst of trying to pilot a deployment of Microsoft Lync (mainly for non-voice collaboration, specifically IM) and integrate it with our Asterisk (11.6.0 if it matters) deployment and a “everything in one place” tool when people are out of the office. I have everything on the voice side playing nice from the Lync side (Lync-Lync, Lync-Asterisk, Lync-Asterisk-PSTN) but I can’t get calls from Asterisk-Lync passing. I think the root issue is Lync demands that the “line URI” be entered in a E.164 normalized format, and further specifies that if an extension is specified it should be entered as ;ext=. So, e.g. when I have myself set up in LYNC my Line URI is entered as “tel:+144044911100;ext=1407+144044911100;ext=1407”. If I try feeding that into an Asterisk DIAL() using any format I can think of (specific examples below) the call fails and the following is logged to console; it looks like Asterisk is dropping the “;ext=”… == Using SIP RTP CoS mark 5 -- Executing [1407@yyy:1] Dial(xx, SIP/lync/ +14404491100) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/lync/+14404491100 -- Got SIP response 485 Ambiguous back from IP address and port of Lync mediation server == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel ' xx' status is 'CHANUNAVAIL' On the other hand, if I change my line URI to a “random” and unused in Lync E.164 number without an extension and change the DIAL() to reflect that number… the call succeeds, so it seems like I’ve narrowed it down to just needing to figure out how to properly pass the extension to Lync. The Googling I turned up didn’t seem too positive (and suggested using an Exchange Unified Messaging auto attendant and forcing the user to redial the extension once connected to the AA was the only alternative for non-DID users) but it seems like it should be relatively simple to bridge (what seems like a very small) gap. Here are the least embarrassing variations on Dial I’ve tried Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above Dial(“SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above Dial(SIP/lync/+14404491100/1407) -- call ‘sits there’ and multiple “sip_xmit of 0x7ffab40891e0 (len 841) to 0.0.5.127:5060 returned -1: Invalid argument” logged to console Any assistance, is as always very appreciated. Thanks! Lincoln -- Lincoln King-Cliby, CTS, DMC-D Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Silver) V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com Crestron Services Provider -- _ -- 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 Fax detection *11.7
Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912 tel:%5B12345678912@from-sip:1] Answer(SIP/abcde-0016, ) in new stack 0x7fd11404cd00 -- Probation passed - setting RTP source address to 123.456.789.123:17108 -- Executing [12345678912 tel:%5B12345678912@from-sip:2] GotoIf(SIP/abcde-0016, 0?black,1) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:3] Ringing(SIP/abcde-0016, ) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:4] Progress(SIP/abcde-0016, ) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:6] Dial(SIP/abcde-0016, SIP/123SIP/456,30,oxX) in new stack == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 -- Called SIP/200 -- Called SIP/201 -- SIP/123-0018 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/456-0017 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/123-0018 is ringing -- SIP/456-0017 is ringing is that what asterisk is showing during an incoming fax call. It looks like the faxdetection is not working but why? Regards Jakob -- _ -- 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 not receiving call from VPN address
David, It seems to me that Asterisk is not seeing/binding to your VPN interface. You need to debug that first. I would set en explicit bind statement in sip.conf to the VPN interface address and nothing else. Then start your asterisk and watch the log messages. It should confirm that it cannot bind to that address. If it does bind, then try your test again and asterisk should see the SIP packets coming in. -- Technical Support http://www.cellroute.net -- _ -- 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 not receiving call from VPN address
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. You might think Kamailio is transmitting it to Asterisk, however without looking at the actually routing tables on Kamailio you'll never know if it actually made it to Asterisk. Again, we need a pcap trace on both Kamailio and Asterisk, plus what your routes look like (route -n), for a call. It will show us clearly what is happening. This all sounds like a routing issue, so your network admins should be able to help troubleshoot. -- _ -- 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 not receiving call from VPN address
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. You might think Kamailio is transmitting it to Asterisk, however without looking at the actually routing tables on Kamailio you'll never know if it actually made it to Asterisk. Again, we need a pcap trace on both Kamailio and Asterisk, plus what your routes look like (route -n), for a call. It will show us clearly what is happening. This all sounds like a routing issue, so your network admins should be able to help troubleshoot. I finally re-read the complete thread. When are you starting the VPN on your Asterisk server, before or after Asterisk has started? If after, and you are binding to 0.0.0.0, it is likely Asterisk is not actually bound to your tun0 interface. So, for a test, explicitly have asterisk listen only on the tun0 interface, retry your call. Or setup your tunnel, then stop Asterisk and start it again, that should cause it to bind properly. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Fax detection *11.7
I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, ) in new stack 0x7fd11404cd00 -- Probation passed - setting RTP source address to 123.456.789.123:17108 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016, 0?black,1) in new stack -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, ) in new stack -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, ) in new stack -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in new stack -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016, SIP/123SIP/456,30,oxX) in new stack == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 -- Called SIP/200 -- Called SIP/201 -- SIP/123-0018 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/456-0017 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/123-0018 is ringing -- SIP/456-0017 is ringing is that what asterisk is showing during an incoming fax call. It looks like the faxdetection is not working but why? Regards Jakob -- _ -- 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 Fax detection *11.7
i already added a Progess() and Wait(5) and it still does not detect faxes. Am 21.01.2014 16:53, schrieb Leandro Dardini: I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de mailto:ja...@j-mb.de Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912 tel:%5B12345678912@from-sip:1] Answer(SIP/abcde-0016, ) in new stack 0x7fd11404cd00 -- Probation passed - setting RTP source address to 123.456.789.123:17108 -- Executing [12345678912 tel:%5B12345678912@from-sip:2] GotoIf(SIP/abcde-0016, 0?black,1) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:3] Ringing(SIP/abcde-0016, ) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:4] Progress(SIP/abcde-0016, ) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in new stack -- Executing [12345678912 tel:%5B12345678912@from-sip:6] Dial(SIP/abcde-0016, SIP/123SIP/456,30,oxX) in new stack == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 -- Called SIP/200 -- Called SIP/201 -- SIP/123-0018 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/456-0017 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/123-0018 is ringing -- SIP/456-0017 is ringing is that what asterisk is showing during an incoming fax call. It looks like the faxdetection is not working but why? Regards Jakob -- _ -- 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 Fax detection *11.7
Please paste the actual code. First has to be the Wait and then any other thing. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de i already added a Progess() and Wait(5) and it still does not detect faxes. Am 21.01.2014 16:53, schrieb Leandro Dardini: I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, ) in new stack 0x7fd11404cd00 -- Probation passed - setting RTP source address to 123.456.789.123:17108 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016, 0?black,1) in new stack -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, ) in new stack -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, ) in new stack -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in new stack -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016, SIP/123SIP/456,30,oxX) in new stack == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 -- Called SIP/200 -- Called SIP/201 -- SIP/123-0018 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/456-0017 connected line has changed. Saving it until answer for SIP/abcde-0016 -- SIP/123-0018 is ringing -- SIP/456-0017 is ringing is that what asterisk is showing during an incoming fax call. It looks like the faxdetection is not working but why? Regards Jakob -- _ -- 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] DUNDI or ENUM or ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 20/01/2014 12:03, Jean-Denis Girard a écrit : Hi list, I'm looking for the best / recommended solution for automatic discovery of phone numbers for a multiple Asterisk system. This would be for an administration, with many branches (~30), but a common infrastructure (DNS, LDAP). Most branches would have Asterisk servers for various reasons (location, administrative). All contacts would be in LDAP, and Asterisk servers would have DNS entries. The problem is contacting other Asterisk without setting static routes in dialplan. I think DUNDI would be ideal, but is it still recommended for new installations or is it deprecated? dundi.com is dead, and redirects to the profile page on Digium website (https://my.digium.com/en/users/viewprofile/). ENUM could be another solution. What would you suggest? No recommendation ? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlLep2QACgkQuu7Rv+oOo/h1YwCgnhs5Pioo0vr5wuWB4yZeDVuJ S+oAnj1GGr7JXtc3wDVyc4wOSN5GZZcw =0O+Z -END PGP SIGNATURE- -- _ -- 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] What is dahdi.auto_assigned_spans and why should you care? (II)
Hi again, Some other things missing from the previous message: setting E1/T1/[J1?] and upgrading. E1/T1 - an E1/T1/J1 should be configured to be either E1 or T1.There are a number of ways to set this. Specifically: quite a few different ways. With the new DAHDI initialization scheme we also provide a new standard method[X] for setting the type (E1/T1/J1) of the span. If you set dahdi.auto_assign_spans=0, the spans are not assigned automatically. Before assigning them, the user may set the spantype attribute[K]. This user is normally the udev hook script /usr/share/dahdi/dahdi_handle_device which will run 'dahdi_span_types set' for the span before assigning the spans of the device[C]. There are several sensible values to put in that file: 1. Just not have it. In this case you fall back to the previous settings of your drivers. 2. All spans are E1: Just include the following line: * *:E1 (If you want T1 or J1. put that type instead) 3. Anything more complex: it's easy to do that. you can start with: dahdi_span_types dumpconfig and set the specific spans there to be E1 or T1. Note that the configuration lines are read in order[D] and hence if you have a default (* *:T1) line and some more specific lines (id1234 [13]:E1), the default must come first. See the sample span-types.conf[4] In 2.8 dahdi_genconf generates span-types.conf by default with lines for each existing Astribank. This has turned out to be a bad idea. In 2.9 merely running 'dahdi_genconf' will not generate span-types.conf . You'll need to explicitly use 'dahdi_genconf spantypes' . And the option you'll normally use is: dahdi_genconf --tdm-type=J1 Upgrades You have on your system a version of DAHDI in which dahdi.auto_assign_spans is set to 1. Now you upgrade it to a version in which auto_assign_spans is set to 0. What happens? Workaround ~~ There is the workaround of setting 'options dahdi auto_assign_spans=1' in a file in /etc/modprobe.d . It will work, but we're trying to avoid that. Planning Ahead ~~ If you plan ahead, you can run before the upgrade: dahdi_genconf assignedspans which is a glorified way of running: dahdi_span_assignments dumpconfig /etc/dahdi/assigend-spans.conf However this will not work for some older versions of DAHDI (as the interface may not be there, is is not complete). And besides, why plan ahead? Just Upgrade So, what happens if If you just load the system (make sure that there's no existing /etc/dahdi/assigned-spans.conf), spans get registered in the same order as they appear. So basically you should see once that the system started properly, and then run: dahdi_genconf assignedspans This will be used for the next time the module is loaded, so there is no need to run anything afterwards to apply the changes. I wonder if this fits in a postinst script of a package of dahdi kernel modules. Astribanks ~~ For Astribanks the upgrade is trickier: Astribanks have their own mechanism that was intended to provide stable channel and span numbers in DAHDI: just before registering all the Astribanks with DAHDI they would wait and registration was done by a specific order (so it worked well s long as noon was added or removed). The configuration is a list of Astribanks in /etc/dahdi/xpp_order . With the upgrade the Astribanks will no longer wait. Thus the Just Upgrade method will not work[2]. This problem, like any other problem in DAHDI, can be fixed by adding another layer of shell scripts. Specifically in case the startup scripts notice that: * The system has Astribanks * There is an xpp_order file and is not completely remmed-out * There is no assigned-spans.conf we run a script that unassigns all devices, and re-assigns them by the order written in the xpp_order file. And write a notice to the console that asks the user to kindly run 'dahdi_genconf assignedspans'. So all's well here as well. Temporarily Set auto_assign_spans=1 ~~~ The following script unloads dahdi and loads it temporarily with auto_assign_spans set to 1 to generate configuration: #!/bin/sh set -e # Make it the last, in case we have those setting elsewhere: modprobe_conf=/etc/modprobe.d/zz_xpp_order_upgrade.conf echo options dahdi auto_assign_spans=1 $modprobe_conf /etc/init.d/dahdi restart dahdi_genconf assignedspans rm $modprobe_conf /etc/init.d/dahdi restart I suspect it won't be needed, but just in case. Conclusion ~~ Upgrading should work. No need to set auto_assign_spans=1 . But please help us test it. [X] XKCD-927 compatible [K] Again, see http://docs.tzafrir.org.il/dahdi-linux/#_devices_bus [C] Again, see http://docs.tzafrir.org.il/dahdi-tools/#_new_devices [D] Still not the case in 2.8.0, will be the case in 2.9.0 [4] http://docs.tzafrir.org.il/dahdi-tools/#_new_devices [2] That is: it will work with probability 1/n!, where n is the number of the Astribanks on the system. --
[asterisk-users] Unknown problem sending outbound fax
All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-0074): Buffer overflow detected (59 + 134 175) Has anyone ever seen this before? I have the following configuration. udptl.conf: [general] udptlstart = 4000 udptlend = 4999 udptlfecentries = 3 udptlfecspan = 3 use_even_ports = no T38FaxUdpEC = t38UDPFEC udptlchecksums = no sip.conf: faxdetect=yes t38pt_rtp=no t38pt_tcp=no t38pt_udptl=yes,fec t38pt_usertpsource=yes Any help at all would be greatly appreciated. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 mailto:f...@voipbusiness.us supp...@voipbusiness.us -- _ -- 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 to provision asterisk's phonebook to Polycom vVX310's
Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley -- _ -- 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] how to provision asterisk's phonebook to Polycom vVX310's
On 01/21/2014 01:55 PM, Stanley van Dijk wrote: Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley Even the old ones could view a webpage. Have a script read the Mysql DB/users table data, then output in XML. The newer ones can output in HTML5. This solution is auto updated when you add GUI users. Or you could maintain a static directory, but this is not good for a large office. The maintenance is impossible. Polycom used to charge for LDAP directory access, this might be free now? Or maybe I dreamt about free LDAP while reading a release note. :) Adrian -- _ -- 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] how to provision asterisk's phonebook to Polycom vVX310's
On Tuesday, January 21, 2014 12:32 PM, Adrian Serafini wrote: On 01/21/2014 01:55 PM, Stanley van Dijk wrote: Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Even the old ones could view a webpage. Have a script read the Mysql DB/users table data, then output in XML. The newer ones can output in HTML5. This solution is auto updated when you add GUI users. ...or, if you use res_phoneprov, Asterisk can auto-generate a static XML phone directory from a template during run-time (whenever the file is requested). I'm not sure how one would go about setting this up if they are using FreePBX, however...I use res_phoneprov in combination with Asterisk-GUI and it works great. Polycom used to charge for LDAP directory access, Pretty sure you're correct, and that this hasn't changed (AFAIK). -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- 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 not receiving call from VPN address
Hi Andres, Thanks for the idea. We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. On 22 January 2014 01:40, Andres and...@telesip.net wrote: David, It seems to me that Asterisk is not seeing/binding to your VPN interface. You need to debug that first. I would set en explicit bind statement in sip.conf to the VPN interface address and nothing else. Then start your asterisk and watch the log messages. It should confirm that it cannot bind to that address. If it does bind, then try your test again and asterisk should see the SIP packets coming in. -- Technical Support http://www.cellroute.net -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 not receiving call from VPN address
Hi Paul, Thanks, we did try restarting Asterisk after the VPN was up but that didn't solve the issue either. On 22 January 2014 02:55, Paul Belanger paul.belan...@polybeacon.comwrote: On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. You might think Kamailio is transmitting it to Asterisk, however without looking at the actually routing tables on Kamailio you'll never know if it actually made it to Asterisk. Again, we need a pcap trace on both Kamailio and Asterisk, plus what your routes look like (route -n), for a call. It will show us clearly what is happening. This all sounds like a routing issue, so your network admins should be able to help troubleshoot. I finally re-read the complete thread. When are you starting the VPN on your Asterisk server, before or after Asterisk has started? If after, and you are binding to 0.0.0.0, it is likely Asterisk is not actually bound to your tun0 interface. So, for a test, explicitly have asterisk listen only on the tun0 interface, retry your call. Or setup your tunnel, then stop Asterisk and start it again, that should cause it to bind properly. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 not receiving call from VPN address
(Please don't top-post.) On Wed, 22 Jan 2014, David Cunningham wrote: We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. Have you confirmed via 'netstat' (or some other system level toop) that Asterisk is actually listening to UDP port 5060 on the VPN IP address? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 not receiving call from VPN address
On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote: (Please don't top-post.) On Wed, 22 Jan 2014, David Cunningham wrote: We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. Have you confirmed via 'netstat' (or some other system level toop) that Asterisk is actually listening to UDP port 5060 on the VPN IP address? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 Hi Steve, When we have bindport = 172.x.x.14 then netstat -udpln shows the following. When bindport is 0.0.0.0 then netstat shows it listening on 0.0.0.0 as you'd expect. udp0 0 172.x.x.14:50600.0.0.0:* 18114/asterisk -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] AMI version to Asterisk version mapping
Is there a mapping of AMI versions to Asterisk versions? eg: AMI 1.0 = Ast 1.4 AMI 1.1 = Ast 1.6 etc... -- _ -- 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] core show channels truncates channel names?
When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data) IAX2/FONEMITEL123456 1296197222@entryhomemailto:1296197222@entryhome Ringing AppDial((Outgoing Line)) SIP/200-093e4998s@macro-dialexternalmailto:s@macro-dialexternal Ring Dial(IAX2/FONEMITEL1234567890/ 2 active channels 1 active call How can I get the full username of all active channels? (I realize I can use the AMI but trying to avoid that) Note that the above output is generated on an Ast 1.4 system -- _ -- 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] core show channels truncates channel names?
On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis mdup...@ocg.ca wrote: When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data) IAX2/FONEMITEL123456 1296197222@entryhome Ringing AppDial((Outgoing Line)) SIP/200-093e4998s@macro-dialexternal Ring Dial(IAX2/FONEMITEL1234567890/ 2 active channels 1 active call How can I get the full username of all active channels? (I realize I can use the AMI but trying to avoid that) Note that the above output is generated on an Ast 1.4 system Use core show channels concise. It was intended for script querying of the channels. However, you really should AMI actions where possible as they are not likely to change from version to version where CLI commands can and do change. Richard -- _ -- 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 not receiving call from VPN address
On Tue, Jan 21, 2014 at 5:18 PM, David Cunningham dcunning...@voisonics.com wrote: On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote: (Please don't top-post.) On Wed, 22 Jan 2014, David Cunningham wrote: We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. Have you confirmed via 'netstat' (or some other system level toop) that Asterisk is actually listening to UDP port 5060 on the VPN IP address? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 Hi Steve, When we have bindport = 172.x.x.14 then netstat -udpln shows the following. When bindport is 0.0.0.0 then netstat shows it listening on 0.0.0.0 as you'd expect. udp0 0 172.x.x.14:50600.0.0.0:* 18114/asterisk -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 At this point in time, you'll need to show us a .pcap on the Asterisk box, when you make a call to it via Kamailio. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Fax detection *11.7
On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger ja...@j-mb.de wrote:
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts: 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels: 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
Digium T.38
Licensed Channels: 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0
so that should be ok.
The corresponding dialplan section starts with
[from-sip]
include = inbound
[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)
in the sip.conf i specified
[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes
...
[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912
but all i can see if i try to send a testfax is
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing
Don't expect T.30 over SIP to be reliable. If you need fax, you should
be using T.38. Your codec is likely the issue.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
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Re: [asterisk-users] Asterisk not receiving call from VPN address
At this point in time, you'll need to show us a .pcap on the Asterisk box, when you make a call to it via Kamailio. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Hi Paul, Thanks for the reply. What are you looking for in the PCAP, that isn't in the tcpdump earlier in the thread? I just want to make sure we gather the information required. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 Fax detection *11.7
Hello,
Perhaps you need to have directmedia=no set for the channel, the call
doesn't appear to have been answered hence asterisk won't be able to
hear any tones to determine for itself if the call is an incoming fax.
Larry.
On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
Digium T.38
Licensed Channels : 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0
so that should be ok.
The corresponding dialplan section starts with
[from-sip]
include = inbound
[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)
in the sip.conf i specified
[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes
...
[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912
but all i can see if i try to send a testfax is
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing
Any hints why thats not working?
Best Regards Jakob
--
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Re: [asterisk-users] Asterisk Fax detection *11.7
Sorry, I missed the line showing the call had been answered.
On 22/01/2014 8:11 AM, Larry Moore wrote:
Hello,
Perhaps you need to have directmedia=no set for the channel, the call
doesn't appear to have been answered hence asterisk won't be able to
hear any tones to determine for itself if the call is an incoming fax.
Larry.
On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
Digium T.38
Licensed Channels : 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0
so that should be ok.
The corresponding dialplan section starts with
[from-sip]
include = inbound
[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)
in the sip.conf i specified
[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes
...
[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912
but all i can see if i try to send a testfax is
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing
Any hints why thats not working?
Best Regards Jakob
--
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Re: [asterisk-users] Asterisk not receiving call from VPN address
On 1/21/14, 4:38 PM, David Cunningham wrote: Hi Andres, Thanks for the idea. We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. Ok, that is progress though. At this point we know that the OS is receiving the packet and Asterisk is listening on that interface and port. I know you already removed the firewall so that would not be the issue. My next guess is Asterisk is looking at the packet and dropping it because it believes it is not meant for it (Kamalio config issue), so try a simple test to confirm this. Just configure a remote IP phone/softphone via that same VPN interface to simulate a remote SIP endpoint. If the SIP phone works fine but Kamalio does not, that will clearly tell you where the problem lies. On 22 January 2014 01:40, Andres and...@telesip.net mailto:and...@telesip.net wrote: David, It seems to me that Asterisk is not seeing/binding to your VPN interface. You need to debug that first. I would set en explicit bind statement in sip.conf to the VPN interface address and nothing else. Then start your asterisk and watch the log messages. It should confirm that it cannot bind to that address. If it does bind, then try your test again and asterisk should see the SIP packets coming in. -- Technical Support http://www.cellroute.net -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- Technical Support http://www.cellroute.net -- _ -- 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] qualify=yes outboundproxy
I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the host= IP, instead of the outboundproxy= IP as it should, verified with tcpdump. I've done a search of the mailing list and didn't turn up anything relevant. Is there a setting I'm missing, is this a bug, or just something that won't work. It seemed appropriate to check here before I filed a bug report! Best Regards, Nicholas Lemberger -- _ -- 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] qualify=yes outboundproxy
Are you absolutely sure you need to use the outboundproxy setting rather than using a peer/friend? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Lemberger Sent: Tuesday, January 21, 2014 7:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] qualify=yes outboundproxy I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the host= IP, instead of the outboundproxy= IP as it should, verified with tcpdump. I've done a search of the mailing list and didn't turn up anything relevant. Is there a setting I'm missing, is this a bug, or just something that won't work. It seemed appropriate to check here before I filed a bug report! -- _ -- 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] Register = plain text password
Hello, Is there anyway to encrypt or scramble a bit the secret used to register with a provider? Im talking about the register = fromuser@fromdomain:secret@host directive in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf This clever dude modified the code back in 1.4: http://www.oneharding.com/voip/asterisk_md5_register.html I imagine that so many years later, and now with the implementation of pjsip this secret could be better protected? It is very unsafe to keep the accounts password right out there. Any ideas? *José Pablo Méndez * -- _ -- 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] Meetme Show Activity in Minus
Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using meetme list command it shows Minus in activity column. Any Idea. meetme list Conf Num PartiesMarked Activity Creation Locked 54682 0002 N/A00:01:31 Dynamic No 62649 0003 N/A00:04:14 Dynamic No *52633 0002 N/A-6:-56:-48 Dynamic No 89737 0001 N/A-6:-40:-42 Dynamic No 89932 0002 N/A-6:-39:-20 Dynamic No 65393 0002 N/A-6:-33:-17 Dynamic No * -- Chandrakant Solanki -- _ -- 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] Meetme Show Activity in Minus
Solved On Wed, Jan 22, 2014 at 12:44 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using meetme list command it shows Minus in activity column. Any Idea. meetme list Conf Num PartiesMarked Activity Creation Locked 54682 0002 N/A00:01:31 Dynamic No 62649 0003 N/A00:04:14 Dynamic No *52633 0002 N/A-6:-56:-48 Dynamic No 89737 0001 N/A-6:-40:-42 Dynamic No 89932 0002 N/A-6:-39:-20 Dynamic No 65393 0002 N/A-6:-33:-17 Dynamic No * -- Chandrakant Solanki -- _ -- 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] Solution to connect an audio system to MeetMe
Hello, Le 16/01/2014 14:20, Darryl Moore a écrit : Yup. That's what i do. The CLI version of linphone set to autoanswer, with the audio jacks tied to our exernal sound system. Works well. The echo cancellation in linphone helps a lot for speakerphones. Indeed, works well :-) Thanks. On Jan 16, 2014 7:51 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: Hi list, I have a customer which will organize a conference in a big meeting room which has a sound system. He would like to connect this sound system to a MeetMe room so participant in the MeetMe can act as if they where on site. My idea is to take a barbone or Notebook, connect it to the sound system using the soundcard and run a softphone on it. Does some of you already have success in such a setup? Which solution did you implement? Any ideas are welcome :-) -- Daniel -- _ -- 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
