Re: [asterisk-users] Paging systems?
No. The SNOM PA1 are headless SIP clients which you configure in auto answer and connect to an amplifier to drive PA speakers. The phones are where you make the announcements from. On Thu, Mar 21, 2019, 5:07 PM Antony Stone, < antony.st...@asterisk.open.source.it> wrote: > On Thursday 21 March 2019 at 21:59:51, Darryl Moore wrote: > > > For a paging system? No you don't. A number of SNOM PA1's and a few > > grandstream phones and you're golden. > > Are you suggesting using standard telephones (presumably in auto-answer > speakerphone mode) as paging devices? > > Depending on the environment, it can work very well (quiet office) or not > at all > (noisy workshop, large factory floor). > > > If you do need FXO or FXS, they are just as easy to setup as well, and > there > > are lots to choose from. > > > > On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis, > wrote: > > > You need more than an ATA. You need something with an FSO and FXO. I’ve > > > used Linksys/SPA3102-3.3.6 and been happy with it. > > > > > > *From:* asterisk-users *On > > > Behalf Of *Sebastian Nielsen > > > *Sent:* Thursday, March 21, 2019 3:01 PM > > > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' < > > > asterisk-users@lists.digium.com> > > > *Subject:* Re: [asterisk-users] Paging systems? > > > > > > How did the page system answer the call when it was used with the > analog > > > system? > > > > > > You could propably ”fake” those signals from inside asterisk, and cause > > > it to answer. > > > > > > *Från:* asterisk-users *För > > > *Michael Munger > > > *Skickat:* den 21 mars 2019 20:00 > > > *Till:* asterisk-users@lists.digium.com > > > *Ämne:* [asterisk-users] Paging systems? > > > > > > Does anyone have an (overhead) paging system that they like that works > > > with SIP? > > > > > > We’ve got a client with an old paging system that (supposedly) just > takes > > > an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it > > > doesn’t auto-answer the call, so paging never happens. > > I'm still intrigued to know how this really was plugged in and how it > operated. > > > Antony. > > -- > All generalisations are inaccurate. > >Please reply to the > list; > please *don't* CC > me. > > -- > _ > -- 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 -- _ -- 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] Paging systems?
For a paging system? No you don't. A number of SNOM PA1's and a few grandstream phones and you're golden. If you do need FXO or FXS, they are just as easy to setup as well, and there are lots to choose from. On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis, wrote: > You need more than an ATA. You need something with an FSO and FXO. I’ve > used Linksys/SPA3102-3.3.6 and been happy with it. > > > > > > > > *From:* asterisk-users *On > Behalf Of *Sebastian Nielsen > *Sent:* Thursday, March 21, 2019 3:01 PM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' < > asterisk-users@lists.digium.com> > *Subject:* Re: [asterisk-users] Paging systems? > > > > How did the page system answer the call when it was used with the analog > system? > > You could propably ”fake” those signals from inside asterisk, and cause it > to answer. > > > > *Från:* asterisk-users *För *Michael > Munger > *Skickat:* den 21 mars 2019 20:00 > *Till:* asterisk-users@lists.digium.com > *Ämne:* [asterisk-users] Paging systems? > > > > Does anyone have an (overhead) paging system that they like that works > with SIP? > > > > We’ve got a client with an old paging system that (supposedly) just takes > an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it > doesn’t auto-answer the call, so paging never happens. > > > > > > Michael J. Munger, dCAP, MCPS, MCNPS, MBSS > > *Microsoft Certified Professional* > > *Microsoft Certified Small Business Specialist* > > *Digium Certified Asterisk Professional* > > *High Powered Help, Inc.* > > p: > > 678-905-8569 > > w: > > hph.io e: m...@hph.io > > > > > -- > _ > -- 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 -- _ -- 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] Paging systems?
I've used SNOM PA1 before with good success. On Thu, Mar 21, 2019, 2:59 PM Michael Munger, wrote: > Does anyone have an (overhead) paging system that they like that works > with SIP? > > > > We’ve got a client with an old paging system that (supposedly) just takes > an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it > doesn’t auto-answer the call, so paging never happens. > > > > > > Michael J. Munger, dCAP, MCPS, MCNPS, MBSS > > *Microsoft Certified Professional* > > *Microsoft Certified Small Business Specialist* > > *Digium Certified Asterisk Professional* > > *High Powered Help, Inc.* > > p: > > 678-905-8569 > > w: > > hph.io e: m...@hph.io > > > > > -- > _ > -- 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 -- _ -- 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] RS485 Audio device
We do something like this, however we have two pairs of wires. One pair is RS-485 for control running at 9600 baud. The other pair is baseband audio which we control through relays on our intercoms. I can't imaging trying to transmit digitally encoded audio over an RS485 network. There are just too many issues with such a setup. cheers, darryl On 2016-11-02 03:46 PM, Jerry Geis wrote: Hi All, The reason for the question was simply that the customer desired some solution called an "AOR" or Area of refuge - I think it was. Basically a call button, microphone and speaker to hear back with the kicker being "a long distance" the solution has to run. RS485 is like 4000 feet. There are solutions our there apparently that are not built on asterisk - so I was just trying to find a solution that potentially worked with asterisk. Thanks! Jerry -- _ -- 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] Delay after Answer
I've seen this sort of thing where a DNS server is programmed in resolv.conf but is not accessible over the network. Threads get blocked, and you have to wait for the DNS query to timeout. On 16-06-07 10:48 AM, Brent Davidson wrote: I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium TDM400 analog card with 2 incoming analog PSTN lines When a call comes in, the system answers, IVR plays, caller dials an extension, Snom 300 rings, handset picked up. Caller continues to hear ringing for another 7 to 10 seconds. Answerer hears a click, a quick burst of audio, then silence, then another click and audio is engaged. I have tried both SIP and RTP debugging and there are absolutely no messages indicating any timeout or retransmit. I am at a total loss. In the past I've always been able to find an answer to issues like this on my own, but this time I just don't know. I was even beginning to suspect the network switch might be bad, but pinging between the server and the phones shows no packet loss and 0.969ms average response time. What am I missing*?* Thanks, Brent Davidson* * -- _ -- 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] SIP port blocking
Hey all. This isn't directly an Asterisk question, but it is Asterisk related because I am using SIP on asterisk. The last couple of days I found that our asterisk box was having all packets originating from port 5060 being blocked. If I moved my SIP port to any other port I could register and place calls, leaving it on 5060 I can do neither. Also if I ran tcpdump on both ends of my truck connection. I could see all packets arriving at the other end ONLY when they were not originating from port 5060. The next question was where was it being blocked. running traceroute yielded the following: root@1940IronStone:~# traceroute -z 1000 -A -U -p 5060 --sport=5060 70.xx.xx.200 traceroute to 70.xx.xx.200 (70.xx.xx.200), 30 hops max, 60 byte packets 1 192.168.1.1 (192.168.1.1) [*] 3.837 ms 5.282 ms 6.280 ms 2 64.230.199.2 (64.230.199.2) [AS577] 9.690 ms * * 3 64.230.232.177 (64.230.232.177) [AS577] 24.936 ms * * 4 agg2-toronto63_xe5-1-0.net.bell.ca (64.230.156.178) [AS577] 40.235 ms * * 5 lns9-toronto63_GE1-0_101.net.bell.ca (64.230.103.145) [AS577] 10.382 ms * * 6 * * * 7 * * * 8 * * * 9 * * * Notice the second and third packet at each hop after the first router all timeout. Even when I put a long delay between packets. Looking further, I find the same response no matter what source port I use. It appears any UDP packet stream from the same port is being blocked. I don't see this behaviour if I allow traceroute to use random source ports for each packet, and I don't see this on other networks. traceroute -A -U -p 5060 70.xx.xx.200 traceroute to 70.xx.xx.200 (70.xx.xx.200), 30 hops max, 60 byte packets 1 192.168.1.1 (192.168.1.1) [*] 62.783 ms 62.759 ms 62.743 ms 2 64.230.199.2 (64.230.199.2) [AS577] 66.565 ms 66.550 ms 66.587 ms 3 64.230.232.177 (64.230.232.177) [AS577] 66.488 ms 66.487 ms 66.535 ms 4 agg2-toronto63_xe5-1-0.net.bell.ca (64.230.156.178) [AS577] 66.521 ms 66.510 ms 66.552 ms Has anybody seen anything like this before? I'm going to send this to the ISP, but I thought I'd find out if anybody else had ever run into it. Thanks, Darryl -- _ -- 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] Peer is UNREACHABLE
Ahh. Seen that before! That suggests to me that you don't have your sip.conf records setup right. What's your sip.conf look like? On 15-05-28 04:51 PM, Luca Bertoncello wrote: Kevin Larsen kevin.lar...@pioneerballoon.com schrieb: The phone you gave your wife is really old. Are you sure it supports SIP OPTIONS? Can you make a call in or out to it? If you can, it is more likely that it just doesn't support that and you can't use a qualify statement. No, I'm not sure. And no, I can't make any call, right now... At least, not connected to my Asterisk... If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but NOT my phone connected on my Asterisk, using the proxy. I can see that in the log: [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have 1234, digest has luca [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device Test1 sip:1234@172.16.34.132;tag=as6dd12e05 Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Peer is UNREACHABLE
I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a call from it?
On 15-05-28 05:09 PM, Luca Bertoncello wrote:
Darryl Moore dar...@moores.ca schrieb:
Ahh. Seen that before! That suggests to me that you don't have your
sip.conf records setup right.
What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register = 004935:MYSECRET@pbxluca/004935
register = 0049351222:MYSECRET@pbxfax/0049351222
register = 0049351333:MYSECRET@pbxanika/0049351333
register = 44:MYSECRET@messagenet/44
[pbxluca]
type=peer
defaultuser=004935
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=004935
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxfax]
type=peer
defaultuser=0049351222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=0049351222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxanika]
type=peer
defaultuser=0049351333
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=anika_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=0049351333
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[messagenet]
type=peer
defaultuser=44
secret=MYSECRET
dtmfmode=rfc2833
host=sip.messagenet.it
context=messagenet_incoming
outboundproxy=sip.messagenet.it
port=5061
fromuser=44
fromdomain=sip.messagenet.it
usereqphone=yes
canreinvite=no
insecure=invite
Here my extensions.conf:
[stdexten]
include = luca_incoming
include = fax_incoming
include = anika_incoming
include = messagenet_incoming
[luca_incoming]
exten = _004935,1,Verbose(2,Call for Luca)
exten = _004935,n,Dial(SIP/004935)
exten = _004935,n,Hangup
[fax_incoming]
exten = _0049351222,1,Verbose(2,Call for FAX)
exten = _0049351222,n,Dial(SIP/0049351222)
exten = _0049351222,n,Hangup
[anika_incoming]
exten = _0049351333,1,Verbose(2,Call for Anika)
exten = _0049351333,n,Dial(SIP/0049351333)
exten = _0049351333,n,Hangup
[messagenet_incoming]
exten = _44,1,Verbose(2,Call from Messagenet)
exten = _44,n,Dial(SIP/004935)
exten = _44,n,Hangup
[myproxy]
exten = _X.,1,Verbose(2,Call from ${CALLERID(num)} to ${EXTEN})
exten = _X.,n,GotoIf($[${CALLERID(num)} = 004935]?dialluca)
exten = _X.,n,GotoIf($[${CALLERID(num)} = 0049351222]?dialfax)
exten = _X.,n,GotoIf($[${CALLERID(num)} = 0049351333]?dialanika)
exten = _X.,n,Dial(SIP/pbxluca/${EXTEN},30,r)
exten = _X.,n,Hangup
exten = _X.,n(dialluca),Verbose(2,Outgoing using pbxluca)
exten = _X.,n(dialluca),Dial(SIP/pbxluca/${EXTEN},30,r)
exten = _X.,n,Hangup
exten = _X.,n(dialfax),Verbose(2,Outgoing using pbxfax)
exten = _X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r)
exten = _X.,n,Hangup
exten = _X.,n(dialanika),Verbose(2,Outgoing using pbxanika)
exten = _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r)
exten = _X.,n,Hangup
And here my users.conf:
[004935]
fullname = luca
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/004935
[0049351222]
fullname = fax
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/0049351222
[0049351333]
fullname = anika
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/0049351333
Now I see this: if I call my phone (004935) from Twinkle it works.
If I call it from the phone of my wife, logged in on the same AsteriskNOW of
Twinkle and able to speak with Twinkle, it does NOT work and I see that in the
Log of my Asterisk:
== Using SIP RTP CoS mark 5
[May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have
1234, digest has luca
[May 28 23:05:59] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate
device Test1
Re: [asterisk-users] Peer is UNREACHABLE
I'd start by turning on sip debugging in asterisk sip set debug ip [your_phone_ip] and use tcpdump or wireshark to see what the OS sees tcpdump host [your_phone_ip] and udp port 5060 On 15-05-28 03:58 PM, Luca Bertoncello wrote: Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer '004935111' is now UNREACHABLE! Last qualify: 0 In the CLI I can see: Name/username HostDyn Nat ACL Port Status 004935111/0049351 192.168.200.11 D 5060 UNREACHABLE 004935122/0049351 192.168.200.10 D 5060 OK (17 ms) 004935133 (Unspecified)D 5060 UNKNOWN 1234 (Unspecified)D 5060 UNKNOWN messagenet/1234567890 212.97.59.765061 Unmonitored pbxanika/004935172.16.34.132 5060 Unmonitored pbxfax/0049351333 172.16.34.132 5060 Unmonitored pbxluca/0049351222 172.16.34.132 5060 Unmonitored 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] Asterisk connects to another Test-VM with AsteriskNOW and to the italian provider Messagenet. Can someone suggest me, what can I do? I can send the configuration file, if they are needed. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] originate , callerid
What about this? No patches needed. exten = my6003,1,Set(CALLERID(ALL)=MyCallerID) same = n,Dial(SIP/6003@asterisk) exten = 6003,n,Set(MyCallerID=test12345) exten = 6003,n,Originate(local/my6003,app,meetme,6003,x) On 14-12-25 06:46 AM, Anthony Messina wrote: On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten = 6003,n,Originate(SIP/6003@asterisk,app,meetme,6003,x) I use this patch https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_callerid.patch because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 -A -- _ -- 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] Suspicious routers
Hello list, I have again come across a router which behaves very badly with my IAX2 packets. This time I've documented it and thought I'd share to see if anyone else has seen similar issues. I have two asterisk servers running behind a dlink DI-604 Internet router. Both are trying to use the same IAX account to connect to the same remote asterisk server to place phone calls. Niether register with the remote box. They both only use it to place outgoing calls when the need arises. They do both monitor quality though, and one works while the other does not. Using tcpdump to see the IAX traffic on both machines yields the following. machine #1 15:27:04.325773 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:27:05.327026 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:27:24.325523 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:27:25.326783 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:27:44.324648 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:27:45.325916 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:28:04.324285 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:28:05.325570 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:28:24.323868 IP 192.168.10.97.4569 99.234.117.121.4569: UDP 15:28:25.325132 IP 192.168.10.97.4569 99.234.117.121.4569: UDP As you can see this machine is trying desperately to talk to the remote server, but there is never any response. - machine #2 - 15:27:04.364718 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:27:05.363308 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:27:24.362713 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:27:25.365736 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:27:33.182751 IP 192.168.10.96.4569 99.234.117.21.4569: UDP 15:27:33.205658 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:27:33.205936 IP 192.168.10.96.4569 99.234.117.21.4569: UDP 15:27:44.362897 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:27:45.364978 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:28:04.362375 IP 99.234.117.21.4569 192.168.10.96.4569: UDP 15:28:05.365890 IP 99.234.117.21.4569 192.168.10.96.4569: UDP Oh look here's the responses! It would appear that all the responses to the packets from both machines are being sent to the one machine. - I've seen and suspected this before, and changing the old cheap routers has generally fixed this, but I'm wondering if anyone else has seen this before, and if there are other routers I need to worry about. I don't yet have an automated way to test routers for this, but I'm seriously thinking about coming up with something. cheers, darryl -- _ -- 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] G729 Licensing Revisited - I'm Sorry!
On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com wrote: No such thing as 'free open source g729 license', if you actually read the site: There is regarding the copyright on the code. The fact it is also patent encumbered is a different issue. DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. So, basically you are illegal using them if you didn't pay for them. Not true. He said it was a lab setup. It is totally legit to use patented processes in an evaluation/lab environment. 3) Is there a performance/stability/security gain when using the commercial vs. open source version or vice versa. See above about about open source license. Your comment about open source is irrelevant to performance, stability, and security. WRT these criteria, I would be surprised if there is much of a difference. The free software isn't locked to a mother board, so that might count towards performance by some measures. Now having said that. I agree once you leave the lab environment and decide you need g.729, you will unfortunatly need a licence to keep using it. The real question is: is there really any choice other than Digium for the licence? Due to the dual licensing of the asterisk code, even if you could license the codec elsewhere, you might be violating Digium's OSS license when you don't but their commercial asterisk license. Cheers, Darryl -- _ -- 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
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. On Jan 16, 2014 7:51 AM, Administrator TOOTAI 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 -- _ -- 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] Text to Speech Engine
http://translate.google.com/translate_tts?tl=enq=i always find google translate works well http://translate.google.com/translate_tts?tl=frq=je trouve toujours google translate fonctionne bien On Jan 10, 2014 3:17 PM, Jai Rangi jpra...@gmail.com wrote: Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- 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] Disable peer from AMI
put it in a different context in your dial plan and use a gotoif statement to control the times it is allowed to dial out. you can also redirect it to a prerecorded message whenever someone tries to use it during the 'off' time. no need for anything as brutal as disabling it in sip.conf. On 2013-10-23 12:37 AM, Michelle Dupuis mdup...@ocg.ca wrote: I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI. Is there a command to enable/disable (or perhaps delete/add) a peer via the AMI? I could create code to modify sip.conf and force a reload, but that seems like the wrong approach... -- _ -- 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] is g729 codec free? or under license???
Thank you Steve, and I read a bit more on the web on this subject including your own well reasoned page at http://www.soft-switch.org/patents/index.html However, despite wide acceptance of the patentability of such codecs (unfortunately), whether they are in fact software patents or not appears to be a matter of opinion. The FSF and Fedora both refer to codec patents as being software patents. http://endsoftpatents.org/2011/02/usa-patent-reform-not-enough/ http://fedoraproject.org/wiki/Software_Patents A quick google search of both terms will show that there are a great many people who see codec patents as software patents, so I don't think I am alone there. Though I think I can see how you differentiate them, I'm not sure there are any simple rules on how you draw that line between them, and that is very problematic. Before the advent of powerful personal computers, codec patents would have few societal issues. I think now, the harm caused by these patents are greater than the benefit. Much like the harm caused by current copyright laws out weigh the current benefits. I was disappointed to read that New Zealand's patent reforms did not go as far as to invalidate codec patents, but I do look forward to g.729 entering the PD in 2016ish and joining MP3s which also recently became an unencumbered format in most countries. On Sun, 2013-10-06 at 00:13 +0800, Steve Underwood wrote: On 10/05/2013 11:07 PM, Darryl Moore wrote: [blink] umm... they are software patents. Really? Do you have expert legal opinion on that? I've never seen anyone competent dispute the patentability of applied signal processing. Such patents get issued all over the world. There are a couple of software patents related to G.729, but those are not part of the essential pool of patents, and those are probably US only. -- _ -- 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] is g729 codec free? or under license???
On 2013-10-04 5:36 PM, Steve Underwood ste...@coppice.org wrote: On 10/05/2013 01:32 AM, Darryl Moore wrote: I'll explain. The g.729 compression algorithm is not protected by copyright, though specific instances may be. It is protected by a patent. http://www.sipro.com/G-729.html An open source version is available here: http://asterisk.hosting.lv/ What stops you from using this, or even your own implementation isn't copyright, but patent protection. It is the right to use the patented technology that you are licensing, not the particular copyrighted coded that implements it. The G.729 codec software at http://asterisk.hosting.lv/actually uses a codec implementation copyrighted by Intel. You need to obey their copyright conditions. correct, and for a few hundred dollars you are free to use it as you see fit, without royalties. note that i also said that the patent license applies even on code that you write yourself. Here you will find the various G.729 patents which were all granted in 1996. https://www.itu.int/ITU-T/recommendations/related_ps.aspx?id_prod=3334 I had thought these expired next year because I was thinking patents were only 18 years. Turns out they are now 20 years, so they really do not expire til some time in 2016. My bad. If you use G.729A (which practically everyone does) I think there are one or two patent which run beyond 2016, at least in the US. perhaps. i do not claim to have fully researched either the patents or the protocol. is 729 compatible with 729a? out of curiosity though i will find out more about these other patents. So in countries that honour software patents, you need to have a license until some time in 2016. In countries which do not, you are free to use these open source codes now. What have the essential patents relevant to G.729 got to do with software patents? [blink] umm... they are software patents. cheers. On Fri, 2013-10-04 at 15:55 +0200, Olivier wrote: H, I'm not sure how g729 licence and software patents relate to each other. Regards, Steve -- _ -- 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] is g729 codec free? or under license???
I'll explain. The g.729 compression algorithm is not protected by copyright, though specific instances may be. It is protected by a patent. http://www.sipro.com/G-729.html An open source version is available here: http://asterisk.hosting.lv/ What stops you from using this, or even your own implementation isn't copyright, but patent protection. It is the right to use the patented technology that you are licensing, not the particular copyrighted coded that implements it. Here you will find the various G.729 patents which were all granted in 1996. https://www.itu.int/ITU-T/recommendations/related_ps.aspx?id_prod=3334 I had thought these expired next year because I was thinking patents were only 18 years. Turns out they are now 20 years, so they really do not expire til some time in 2016. My bad. So in countries that honour software patents, you need to have a license until some time in 2016. In countries which do not, you are free to use these open source codes now. cheers. On Fri, 2013-10-04 at 15:55 +0200, Olivier wrote: H, I'm not sure how g729 licence and software patents relate to each other. -- _ -- 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] is g729 codec free? or under license???
Hi guys. I would also add that in countries which do not recognize software patents (New Zealand for example) there is no need to get a license and the codecs can therefore be downloaded from http://asterisk.hosting.lv/ and used freely. In countries where there is ambiguity about certain software patents, (such as Canada or Germany) well you take your chances. Countries that do recognize them (such as the US or Japan) you'd be smart to get a license. Also I do believe the US patent on g729 expires next year anyway, so again you might want to weigh that costs/risks factor too. cheers, darryl On Wed, 2013-10-02 at 09:55 -0400, Bryant Zimmerman wrote: When calling between two g729 client endpoints you do not need any licenses as long as no audio prompts or voicemail is evolved. Also as a sip trunk provider we offer g729 as a source and destination codec this allows you to make calls in and out using g729 (most carrier grade providers offer this option) You really only need to buy the number of g729 licenses that you will need for callers that require simultaneous transcoding. This is when a callers stream in or out will need to be converted to another codec format. This occurs when callers are jumping from say g729 to g711 or g729 to g722, g729 to gsm. If you plan things right and make sure any audio prompts your system is using are recorded in g729 as well as g711 and g722 you will reduce the number of g729 license considerable. Process that use a lot of g729 transcodes. ConfBridge uses g722 so all g729 has to be converted to and from g722 so 10 g729 callers to a confbridge would likely require 10 codecs (**See confbridge trick below). If you have prompts that are not pre-encoded in g729 those would use a transcoder license while playing. Voicemail would require a license as g729 has to be transcoded to one of the storage formats. The real number is based on how you are using your system. ConfBridge Trick - Have seen this used for voicemail as well, Make sure you test when using this method. If you can live with using higher bandwidth to the asterisk switch when using confbridges (endpoints also have to support in call reinvites correctly) you can force endpoints to re-invite to g722 before dropping into the conference bridge. This has the upside of not needing to transcode on the server thus improving performance and reducing g729 license requirements. This comes at the cost of needing higher bandwidth between the client endpoints and the phone. Figure about double the bandwidth when using this method. It may or may not be worth it to you depending on your scenario. Please let us know if this information helps you. Thanks Bryant Zimmerman Sr. Systems Architect Grand Dial Communications, A ZK Tech Inc. Company 616-299-5607 (mobile) 616-855-1030 Ext. 2003 (office) __ From: Don Kelly d...@donkelly.biz Sent: Wednesday, October 2, 2013 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] is g729 codec free? or under license??? In your scenario, all the calls are from endpoints on 181 to endpoints on 183. If the endpoint devices are similar, it seems to me that there should be no need to transcode-you can use a codec common to the endpoints. 729 would not be required. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of s m Sent: Wednesday, October 02, 2013 2:34 AM To: Dominik George Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] is g729 codec free? or under license??? thank you Dominik you help me a lot. and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? please help me to clarify channel concept in my mind. thanks in advance SAM On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi, about g729, you mean if it get free g729 and all my systems (PBXs and routers) use g729 codec for setting a call, call is set without any problem? Yes, if all systems use g729 directly, you are ready to go. - -nik -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid
Re: [asterisk-users] iax packet loss again.
OK list. Just in case anyone cares. I did figure out (sort of) why I was not receiving any IAX packets. http://lists.digium.com/pipermail/asterisk-users/2013-September/280590.html My modules.conf file was the same one I had previously used in 1.8 and it specifically loaded necessary modules instead of using autoload = yes. This was because previously I had been having issues with a few modules and my simple application did not need many, so this was the easy solution. I'm not sure what modules IAX requires in 11 which were not required in 1.8 so as not to choke. (Well other then chan_iax2 of course which I was using.) Any way. If any one else has issues with IAX packets (I've seen a small number) check your modules.conf file and see if you can figure out what undocumented module it is missing. TTFN darryl -- _ -- 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] iax packet loss again.
I saw this thread which is very similar to my issue, though I cannot solve mine with iptables. http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.html Using asterisk 11.5, IAX does not seem to be able to receive any packets. My IP tables looks like this: root@dlaptop:/home/darryl# iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination could it be any simpler with IAX debugging on in asterisk I see this in the console: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00015ms SCall: 00525 DCall: 0 [184.75.215.106:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 00890 DCall: 0 [67.205.74.184:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00010ms SCall: 05381 DCall: 0 [99.245.204.155:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00015ms SCall: 00525 DCall: 0 [184.75.215.106:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 00890 DCall: 0 [67.205.74.184:4569] Notice there are no Rx-Frames, and my peer table looks like this: dlaptop*CLI iax2 show peers Name/UsernameHost Mask Port Status Description voipms/121322_i 184.75.215.106 (S) 255.255.255.255 4569 UNREACHABLE voipms2/121322_ 67.205.74.184 (S) 255.255.255.255 4569 UNREACHABLE /99.245.204.155 (S) 255.255.255.255 4569 UNREACHABLE 3 iax2 peers [0 online, 3 offline, 0 unmonitored] tcpdump can see all the packets though: 17:23:35.840421 IP 184-75-215-106.amanah.com.iax dlaptop-2.local.iax: UDP, length 12 17:23:35.872904 IP 67.205.74.184.iax dlaptop-2.local.iax: UDP, length 12 17:23:36.790984 IP dlaptop-2.local.iax CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax: UDP, length 14 17:23:36.792680 IP CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax dlaptop-2.local.iax: UDP, length 12 17:23:36.814493 IP dlaptop-2.local.iax 184-75-215-106.amanah.com.iax: UDP, length 14 17:23:36.834119 IP dlaptop-2.local.iax 67.205.74.184.iax: UDP, length 14 17:23:36.842537 IP 184-75-215-106.amanah.com.iax dlaptop-2.local.iax: UDP, length 12 17:23:36.877078 IP 67.205.74.184.iax dlaptop-2.local.iax: UDP, length 12 17:23:43.836844 IP dlaptop-2.local.iax CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax: UDP, length 24 17:23:43.838705 IP CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax dlaptop-2.local.iax: UDP, length 65 but my socket buffers are backing up horribly: root@dlaptop:/home/darryl# lsof -n -P -Tq | grep UDP | grep 4569 lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/darryl/.gvfs Output information may be incomplete. asterisk 2575root8u IPv4 334734592 0t0 UDP *:4569 (QR=163904 QS=0) Am i crazy? Is there something as simple as iptables that I missed? Or is there some kind of bug in Asterisk which is being missed? I've only had this issue on two machines which I've compiled 11.5 on. Generally all my production machines are using the stock version 1.8 which is in the Ubuntu 12.04 repository. Unloading and reloading the chan_iax module only has the effect of resetting the receive queue size in lsof. Anyone have any ideas what I could possibly be missing here? Sip works fine by the way. Thanks Darryl -- _ -- 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
