Re: [asterisk-users] issue with NAT
Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens: First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: [TK]D-Fender Note for elfranne's situation, : nat=force_rport,comedia should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it? A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu Hi Tom, you can add a STUN Server in your Linksys/SPA942-6.1.5(a) user agents. read more about STUN at: http://www.voip-info.org/wiki/view/STUN and there is a list of public STUN Server. Regards -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] issue with NAT
Am 03.11.2014 um 13:47 schrieb Rainer Piper: Am 03.11.2014 um 13:28 schrieb Tom Braarup Cuykens: First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: [TK]D-Fender Note for elfranne's situation, : nat=force_rport,comedia should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it? A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu Hi Tom, you can add a STUN Server in your Linksys/SPA942-6.1.5(a) user agents. read more about STUN at: http://www.voip-info.org/wiki/view/STUN and there is a list of public STUN Server. Regards the add path header support in chan_sip could help as well. look at https://issues.asterisk.org/jira/browse/ASTERISK-16884 [Test danes 202] ... ... nat=force_rport,comedia usepath=yes ... ... [test danes 203] ... ... nat=force_rport,comedia usepath=yes ... ... -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] PBX hacked: why hundred of calls to the same number ?
the attacking server changed the destination Number at 18:53 CEST and he is still blocked ... LOL 972597438354 callto:00972597438354 Oct 3 18:53:17 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=00972597438354 callto:00972597438354 Oct 3 19:06:37 server /sbin/kamailio[3978]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=000972597438354 Oct 3 19:19:45 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=972597438354 Oct 3 19:32:59 server /sbin/kamailio[3978]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=*000972597438354 Oct 3 19:46:20 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=100972597438354 Am 03.10.2014 um 14:52 schrieb Rainer Piper: Am 02.10.2014 um 15:40 schrieb Tzafrir Cohen: On Thu, Oct 02, 2014 at 07:52:34AM +0200, Rainer Piper wrote: Is the destination Number like Country Code +972? +972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers] source -http://www.wtng.info/wtng-972-il.html That page is slightly dated. +972 59 XXX are all the numbers in the Palestinian Authority (there are several providers besides Jawall). My SIP Proxy logs all the unauth. INVITEs and I found the a lot calls go to the Country code +972 xxx As a resident of +972 (+972-4), I'll just note that those hack attempts are typically related to PA numbers (+972-59) as rates there are higher. Hi Tzafrir, ok, the page www.wtng.info is not really up to date. here some logs to see the variations of the attempt to dial over my proxy Oct 3 11:23:06 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=00972592910519 callto:00972592910519 Oct 3 11:42:52 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=972592910519 Oct 3 11:53:15 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=700972592910519 Oct 3 12:06:32 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=200972592910519 Oct 3 12:20:04 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=#00972592910519 callto:00972592910519 Oct 3 12:32:53 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=*000972592910519 Oct 3 12:45:35 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=*972592910519 Oct 3 12:57:42 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=900972592910519 Oct 3 13:09:37 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=7700972592910519 Oct 3 13:21:24 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=66600972592910519 Oct 3 13:33:11 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=00972592910519 and the source IP 69.30.254.234 is coming from OrgName:WholeSale Internet, Inc. OrgId: WHOLE-125 Address:324 E. 11th St. Address:Suite 1000 City: Kansas City StateProv: MO PostalCode: 64106 Country:US very strange ;-) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] PBX hacked: why hundred of calls to the same number ?
Hi Eric I like your approach. I think about stateless redirect the bad boy to the NSA- or Pentagon-IVR LOL Am 03.10.2014 um 20:01 schrieb Eric Wieling: We set up our servers to allowguest=yes and autocreatepeer=yes and use a global context setting to point any of those calls to an IVR jail.Attempts stop reasonably quickly. An empty room with an unlocked door is far less interesting than a room with the door locked. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rainer Piper *Sent:* Friday, October 03, 2014 1:53 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ? the attacking server changed the destination Number at 18:53 CEST and he is still blocked ... LOL 972597438354 callto:00972597438354 Oct 3 18:53:17 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=00972597438354 callto:00972597438354 Oct 3 19:06:37 server /sbin/kamailio[3978]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=000972597438354 Oct 3 19:19:45 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=972597438354 Oct 3 19:32:59 server /sbin/kamailio[3978]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=*000972597438354 Oct 3 19:46:20 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=100972597438354 Am 03.10.2014 um 14:52 schrieb Rainer Piper: Am 02.10.2014 um 15:40 schrieb Tzafrir Cohen: On Thu, Oct 02, 2014 at 07:52:34AM +0200, Rainer Piper wrote: Is the destination Number like Country Code +972? +972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers] source -http://www.wtng.info/wtng-972-il.html That page is slightly dated. +972 59 XXX are all the numbers in the Palestinian Authority (there are several providers besides Jawall). My SIP Proxy logs all the unauth. INVITEs and I found the a lot calls go to the Country code +972 xxx As a resident of +972 (+972-4), I'll just note that those hack attempts are typically related to PA numbers (+972-59) as rates there are higher. Hi Tzafrir, ok, the page www.wtng.info http://www.wtng.info is not really up to date. here some logs to see the variations of the attempt to dial over my proxy Oct 3 11:23:06 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=00972592910519 callto:00972592910519 Oct 3 11:42:52 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=972592910519 Oct 3 11:53:15 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=700972592910519 Oct 3 12:06:32 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=200972592910519 Oct 3 12:20:04 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=#00972592910519 callto:00972592910519 Oct 3 12:32:53 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=*000972592910519 Oct 3 12:45:35 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=*972592910519 Oct 3 12:57:42 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=900972592910519 Oct 3 13:09:37 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=7700972592910519 Oct 3 13:21:24 server /sbin/kamailio[7217]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=66600972592910519 Oct 3 13:33:11 server /sbin/kamailio[7218]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=00972592910519 and the source IP 69.30.254.234 is coming from OrgName:WholeSale Internet, Inc. OrgId: WHOLE-125 Address:324 E. 11th St. Address:Suite 1000 City: Kansas City StateProv: MO PostalCode: 64106 Country:US very strange ;-) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de mailto:rai...@xmpp.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123
Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?
Hi Chris, yes ... it is boring ... I stop posting ... ;-) Am 03.10.2014 um 20:11 schrieb Chris Bagnall: On 3/10/14 6:52 pm, Rainer Piper wrote: the attacking server changed the destination Number at 18:53 CEST and he is still blocked ... LOL 972597438354 callto:00972597438354 It's pretty much an everyday occurrence for any internet-connected SIP system these days... Oct 3 19:46:20 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=100972597438354 Many of these attacks come from fairly easily recognised user-agent strings, so if you fancy doing a bit of packet inspection with your firewall, you can block many of these before they get as far as your SIP server(s) themselves. For example, the sipcli scans you listed above can be blocked fairly easily with: iptables -A INPUT -p udp --dport 5060 -m string --algo bm --string sipcli -j DROP (obviously there are overheads to string searching UDP/5060 packets that you'll want to consider, and the above won't work if you're using sipcli legitimately anywhere on your network) Kind regards, Chris -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] PBX hacked: why hundred of calls to the same number ?
just one more ;-) the source IP just changed to 142.0.41.179 OrgName:VolumeDrive OrgId: VOLUM-2 Address:1143 Northern Blvd City: Clarks Summit StateProv: PA PostalCode: 18411 Country:US and the destination Number to 972595632276 callto:00972595632276 Oct 3 20:26:37 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 142.0.41.179 sipcli/v1.8 rm=INVITE aU=null rU=+972595632276 callto:00972595632276 Am 03.10.2014 um 20:15 schrieb Rainer Piper: Hi Chris, yes ... it is boring ... I stop posting ... ;-) Am 03.10.2014 um 20:11 schrieb Chris Bagnall: On 3/10/14 6:52 pm, Rainer Piper wrote: the attacking server changed the destination Number at 18:53 CEST and he is still blocked ... LOL 972597438354 callto:00972597438354 It's pretty much an everyday occurrence for any internet-connected SIP system these days... Oct 3 19:46:20 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=100972597438354 Many of these attacks come from fairly easily recognised user-agent strings, so if you fancy doing a bit of packet inspection with your firewall, you can block many of these before they get as far as your SIP server(s) themselves. For example, the sipcli scans you listed above can be blocked fairly easily with: iptables -A INPUT -p udp --dport 5060 -m string --algo bm --string sipcli -j DROP (obviously there are overheads to string searching UDP/5060 packets that you'll want to consider, and the above won't work if you're using sipcli legitimately anywhere on your network) Kind regards, Chris -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] PBX hacked: why hundred of calls to the same number ?
Am 01.10.2014 um 15:48 schrieb Gokan Atmaca: Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number. For curiosity's sake, I'm wondering why would this happen (dialing the same number over and over) ? Some special numbers generate here and there revenues for callees (and not for callers). Beside sharing interests with the callee that get those revenues, why a hacker would like to dial the same numbers over and over ? In other words, in this case, is looking at callee number a promising path to find hackers ? Is there a bot virus ? Do you IP address restrictions ? I have one SIP Proxy without any outbound trunks/routing and this Proxy is just collecting bad source IPs and bad destination numbers for the database blacklist table and I use this blacklist table in my productive System. On Wed, Oct 1, 2014 at 4:36 PM, Administrator TOOTAI ad...@tootai.net wrote: Le 01/10/2014 11:40, Olivier a écrit : Hi, Hi Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number. For curiosity's sake, I'm wondering why would this happen (dialing the same number over and over) ? Some special numbers generate here and there revenues for callees (and not for callers). Beside sharing interests with the callee that get those revenues, why a hacker would like to dial the same numbers over and over ? callee is also the bad men. Go and buy an 899 number in France, hack PBXS and call your number :-) [...] -- 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 -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] PBX hacked: why hundred of calls to the same number ?
this morning.: Oct 2 07:32:15 server /sbin/kamailio[29866]: NOTICE: script: blocking IP 69.30.254.234 sipcli/v1.8 rm=INVITE aU=null rU=00972597613940 callto:00972597613940 -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] Realtime ERROR
Am 25.09.2014 um 16:24 schrieb Andrew Colin: Hi Guys, I have recently moved my database servers to a different database cluster that runs on haproxy. Every minute or so I get the following error in asterisk MySQL RealTime: Ping failed (2006). Trying an explicit reconnect The strange thing is if I do realtime mysql status It shows as connected just the timer resets. Any idea why this is occurring? Hi Andrew, what balancing algorithm you use in haproxy.cfg ? balance source balance roundrobin or balance leastconn -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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
[asterisk-users] fail2ban and pjsip in asterisk 12 and 13
Hi,
Info !!! not a question !!!
the pjsip logger is different:
[Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c: Request
from '1001 sip:1001@81.20.137.222' failed for '85.25.197.23:5071'
(callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc) - No matching endpoint found
and here the RegEx for fail2ban to catch this log:
|NOTICE.* .*: Request from '.*' failed for 'HOST(:[0-9]{1,5})?' (.*) -
No matching endpoint found
Regards|
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161 callto:004922897167161
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test)
XMPP: rai...@xmpp.soho-piper.de
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13
Am 15.09.2014 um 15:26 schrieb Matthew Jordan:
On Mon, Sep 15, 2014 at 6:21 AM, Patrick Laimbock
patr...@laimbock.com mailto:patr...@laimbock.com wrote:
Hi Rainer,
On 15-09-14 09:07, Rainer Piper wrote:
Hi,
Info !!! not a question !!!
the pjsip logger is different:
[Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c:
Request
from '1001 sip:1001@81.20.137.222
mailto:sip%3A1001@81.20.137.222' failed for
'85.25.197.23:5071 http://85.25.197.23:5071'
(callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc) - No matching
endpoint found
and here the RegEx for fail2ban to catch this log:
|NOTICE.* .*: Request from '.*' failed for
'HOST(:[0-9]{1,5})?' (.*) -
No matching endpoint found
Thanks for sharing. If you use github it would be nice if you
could submit a pull request so that it becomes part of the
Asterisk rules in the next Fail2ban version (0.9.1).
https://github.com/fail2ban/fail2ban/pulls
HTH,
Patrick
Why would you not use the SECURITY log format, which have the exact
same format between chan_sip and chan_pjsip, and have a consistent
format from Asterisk 10+?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
Thanks for security_log = security
Ok ... I switched the
security_log = security
in logger.conf on and I'm going to write a RegEx for Fail2ban.
log sample - security log of wrong password:
[Sep 15 15:51:26] SECURITY[17378] res_security_log.c:
SecurityEvent=ChallengeResponseFailed,EventTV=2014-09-15T15:51:26.126+0200,Severity=Error,Service=PJSIP,EventVersion=1,AccountID=7002,SessionID=80DFFBE5-4C3B-E411-8429-AD5D2362CB3E@192.168.8.10,LocalAddress=IPV4/UDP/178.5.154.91/5072,RemoteAddress=IPV4/UDP/192.168.8.10/6012,Challenge=1410789078/000dd605e4bd1b6dd7488afafafafafaf,Response=8fc17a017a3ac5eea21ca86c6c0f5ee8,ExpectedResponse=
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161 callto:004922897167161
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test)
XMPP: rai...@xmpp.soho-piper.de
--
_
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Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13
Hi Patrick,
github done ;-)
what is HTH ???
Am 15.09.2014 um 13:21 schrieb Patrick Laimbock:
Hi Rainer,
On 15-09-14 09:07, Rainer Piper wrote:
Hi,
Info !!! not a question !!!
the pjsip logger is different:
[Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c: Request
from '1001 sip:1001@81.20.137.222' failed for '85.25.197.23:5071'
(callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc) - No matching endpoint found
and here the RegEx for fail2ban to catch this log:
|NOTICE.* .*: Request from '.*' failed for 'HOST(:[0-9]{1,5})?' (.*) -
No matching endpoint found
Thanks for sharing. If you use github it would be nice if you could
submit a pull request so that it becomes part of the Asterisk rules in
the next Fail2ban version (0.9.1).
https://github.com/fail2ban/fail2ban/pulls
HTH,
Patrick
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test)
XMPP: rai...@xmpp.soho-piper.de
--
_
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Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13
oh ... thanks :-[ Am 15.09.2014 um 17:30 schrieb A J Stiles: (this is not where your reply belongs) On Monday 15 Sep 2014, Rainer Piper wrote: Hi Patrick, github done ;-) what is HTH ??? HTH == Hope That Helps. -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] How to use phpmyadmin to remotely access asterisk mysql database?
Am 11.09.2014 um 10:36 schrieb rafa alfurqan: Hi, thank you for your repplied, As you're on Ubuntu, you can begin with $ sudo apt-get install phpmyadmin i did that, so what i have to do for the configuration in asterisk so i could remote to asterisk database from phpmyadmin? Also, 10.04 is a really old Ubuntu release now, even although it is a Long Term Support one. Consider upgrading to 14.04. You can apt-get dist-upgrade straight from an LTS release to the next LTS release, without needing to go through all the intermediate releases. really appreciate for the advice, i'll do that after i could remote to asterisk database from phpmyadmin. actually i have installed freeradius-server on my ubuntu too, and i could remote the database freeradius from phpmyadmin. is it possible if same phpmyadmin could remote database from freeradius-server and asterisk (they are on same server)? thank you are you sure about allowing remote access to phpmyadmin ??? think about security first !!! I suggest HeidiSQL Client at your Home PC connecting via SSH Tunnel to your remote mySQL listening at localhost only. link to heidiSQL - http://www.heidisql.com/ -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.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] How to use phpmyadmin to remotely access asterisk mysql database?
Am 11.09.2014 um 11:00 schrieb rafa alfurqan: Hi Rainer, are you sure about allowing remote access to phpmyadmin ??? think about security first !!! yes i'm sure coz it's not for commercial, just for my research. I suggest HeidiSQL Client at your Home PC connecting via SSH Tunnel to your remote mySQL listening at localhost only. i just heard HeidiSQL from here, it's like a tool? actually i have no basic for anything about database or voip. HeidiSQL is like phpmyadmin have a look at http://www.heidisql.com/ Features * Free for everyone, OpenSource. * Connect to multiple servers in one window * Connect to servers via commandline http://heidisql.googlecode.com/svn/trunk/out/readme.txt * Connect via SSH tunnel, or pass SSL settings * Create and edit tables http://www.heidisql.com/screenshots.php?which=table_editor, views http://www.heidisql.com/screenshots.php?which=view_editor, stored routines http://www.heidisql.com/screenshots.php?which=stored_routines, triggers http://www.heidisql.com/screenshots.php?which=trigger_editor and scheduled events http://www.heidisql.com/screenshots.php?which=event_editor. * Generate nice SQL-exports http://www.heidisql.com/screenshots.php?which=export_sql, compress these afterwards, or put them on the clipboard. * Export from one server/database directly to another server/database * Manage user-privileges http://www.heidisql.com/screenshots.php?which=usermanager * Import text-files http://www.heidisql.com/screenshots.php?which=import_textfile * Export table rows as CSV, HTML, XML, SQL, LaTeX, Wiki Markup and PHP Array http://www.heidisql.com/screenshots.php?which=export_textfile * Browse and edit table-data using a comfortable grid http://www.heidisql.com/screenshots.php?which=data * Bulk edit tables (move to db, change engine, collation etc.) * Batch-insert ascii or binary files into tables http://www.heidisql.com/screenshots.php?which=insert_files * Write queries with customizable syntax-highlighting and code-completion http://www.heidisql.com/screenshots.php?which=query * Pretty reformat disordered SQL * Monitor and kill client-processes http://www.heidisql.com/screenshots.php?which=host_processlist * Find specific text in all tables of all databases of one server http://www.heidisql.com/screenshots.php?which=find_text_on_server * Optimize and repair tables in a batch manner http://www.heidisql.com/screenshots.php?which=maintenance * Launch a parallel mysql.exe command line window using your current connection settings * And much more http://www.heidisql.com/screenshots.php if you mind, would you give me a tutorial or anything to configure that? thank you so much On Thu, Sep 11, 2014 at 3:46 PM, Rainer Piper rainer.pi...@soho-piper.de mailto:rainer.pi...@soho-piper.de wrote: Am 11.09.2014 um 10:36 schrieb rafa alfurqan: Hi, thank you for your repplied, As you're on Ubuntu, you can begin with $ sudo apt-get install phpmyadmin i did that, so what i have to do for the configuration in asterisk so i could remote to asterisk database from phpmyadmin? Also, 10.04 is a really old Ubuntu release now, even although it is a Long Term Support one. Consider upgrading to 14.04. You can apt-get dist-upgrade straight from an LTS release to the next LTS release, without needing to go through all the intermediate releases. really appreciate for the advice, i'll do that after i could remote to asterisk database from phpmyadmin. actually i have installed freeradius-server on my ubuntu too, and i could remote the database freeradius from phpmyadmin. is it possible if same phpmyadmin could remote database from freeradius-server and asterisk (they are on same server)? thank you are you sure about allowing remote access to phpmyadmin ??? think about security first !!! I suggest HeidiSQL Client at your Home PC connecting via SSH Tunnel to your remote mySQL listening at localhost only. link to heidiSQL - http://www.heidisql.com/ -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de mailto:rai...@xmpp.soho-piper.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 -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP
Re: [asterisk-users] How to use phpmyadmin to remotely access asterisk mysql database?
Am 11.09.2014 um 11:28 schrieb Rainer Piper: Am 11.09.2014 um 11:00 schrieb rafa alfurqan: Hi Rainer, are you sure about allowing remote access to phpmyadmin ??? think about security first !!! yes i'm sure coz it's not for commercial, just for my research. I suggest HeidiSQL Client at your Home PC connecting via SSH Tunnel to your remote mySQL listening at localhost only. i just heard HeidiSQL from here, it's like a tool? actually i have no basic for anything about database or voip. I use HeidiSQL to administrate my 9x Nodes MariaDB Galera 10.0 Multimaster DB Cluster. ;-) *SMILE* I just love this tool ;-) HeidiSQL is like phpmyadmin have a look at http://www.heidisql.com/ Features * Free for everyone, OpenSource. * Connect to multiple servers in one window * Connect to servers via commandline http://heidisql.googlecode.com/svn/trunk/out/readme.txt * Connect via SSH tunnel, or pass SSL settings * Create and edit tables http://www.heidisql.com/screenshots.php?which=table_editor, views http://www.heidisql.com/screenshots.php?which=view_editor, stored routines http://www.heidisql.com/screenshots.php?which=stored_routines, triggers http://www.heidisql.com/screenshots.php?which=trigger_editor and scheduled events http://www.heidisql.com/screenshots.php?which=event_editor. * Generate nice SQL-exports http://www.heidisql.com/screenshots.php?which=export_sql, compress these afterwards, or put them on the clipboard. * Export from one server/database directly to another server/database * Manage user-privileges http://www.heidisql.com/screenshots.php?which=usermanager * Import text-files http://www.heidisql.com/screenshots.php?which=import_textfile * Export table rows as CSV, HTML, XML, SQL, LaTeX, Wiki Markup and PHP Array http://www.heidisql.com/screenshots.php?which=export_textfile * Browse and edit table-data using a comfortable grid http://www.heidisql.com/screenshots.php?which=data * Bulk edit tables (move to db, change engine, collation etc.) * Batch-insert ascii or binary files into tables http://www.heidisql.com/screenshots.php?which=insert_files * Write queries with customizable syntax-highlighting and code-completion http://www.heidisql.com/screenshots.php?which=query * Pretty reformat disordered SQL * Monitor and kill client-processes http://www.heidisql.com/screenshots.php?which=host_processlist * Find specific text in all tables of all databases of one server http://www.heidisql.com/screenshots.php?which=find_text_on_server * Optimize and repair tables in a batch manner http://www.heidisql.com/screenshots.php?which=maintenance * Launch a parallel mysql.exe command line window using your current connection settings * And much more http://www.heidisql.com/screenshots.php if you mind, would you give me a tutorial or anything to configure that? thank you so much On Thu, Sep 11, 2014 at 3:46 PM, Rainer Piper rainer.pi...@soho-piper.de mailto:rainer.pi...@soho-piper.de wrote: Am 11.09.2014 um 10:36 schrieb rafa alfurqan: Hi, thank you for your repplied, As you're on Ubuntu, you can begin with $ sudo apt-get install phpmyadmin i did that, so what i have to do for the configuration in asterisk so i could remote to asterisk database from phpmyadmin? Also, 10.04 is a really old Ubuntu release now, even although it is a Long Term Support one. Consider upgrading to 14.04. You can apt-get dist-upgrade straight from an LTS release to the next LTS release, without needing to go through all the intermediate releases. really appreciate for the advice, i'll do that after i could remote to asterisk database from phpmyadmin. actually i have installed freeradius-server on my ubuntu too, and i could remote the database freeradius from phpmyadmin. is it possible if same phpmyadmin could remote database from freeradius-server and asterisk (they are on same server)? thank you are you sure about allowing remote access to phpmyadmin ??? think about security first !!! I suggest HeidiSQL Client at your Home PC connecting via SSH Tunnel to your remote mySQL listening at localhost only. link to heidiSQL - http://www.heidisql.com/ -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) XMPP: rai...@xmpp.soho-piper.de mailto:rai...@xmpp.soho-piper.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
Re: [asterisk-users] How to use phpmyadmin to remotely access asterisk mysql database?
Am 11.09.2014 um 11:39 schrieb rafa alfurqan: Hi Rainer, okay, thanks for your advice. so i think it would work for freeradius too. I think so ... freeradius DB is mySQL or oracle. Cheers On Thu, Sep 11, 2014 at 4:28 PM, Rainer Piper rainer.pi...@soho-piper.de mailto:rainer.pi...@soho-piper.de wrote: Am 11.09.2014 um 11:00 schrieb rafa alfurqan: Hi Rainer, are you sure about allowing remote access to phpmyadmin ??? think about security first !!! yes i'm sure coz it's not for commercial, just for my research. I suggest HeidiSQL Client at your Home PC connecting via SSH Tunnel to your remote mySQL listening at localhost only. i just heard HeidiSQL from here, it's like a tool? actually i have no basic for anything about database or voip. HeidiSQL is like phpmyadmin have a look at http://www.heidisql.com/ Features * Free for everyone, OpenSource. * Connect to multiple servers in one window * Connect to servers via commandline http://heidisql.googlecode.com/svn/trunk/out/readme.txt * Connect via SSH tunnel, or pass SSL settings * Create and edit tables http://www.heidisql.com/screenshots.php?which=table_editor, views http://www.heidisql.com/screenshots.php?which=view_editor, stored routines http://www.heidisql.com/screenshots.php?which=stored_routines, triggers http://www.heidisql.com/screenshots.php?which=trigger_editor and scheduled events http://www.heidisql.com/screenshots.php?which=event_editor. * Generate nice SQL-exports http://www.heidisql.com/screenshots.php?which=export_sql, compress these afterwards, or put them on the clipboard. * Export from one server/database directly to another server/database * Manage user-privileges http://www.heidisql.com/screenshots.php?which=usermanager * Import text-files http://www.heidisql.com/screenshots.php?which=import_textfile * Export table rows as CSV, HTML, XML, SQL, LaTeX, Wiki Markup and PHP Array http://www.heidisql.com/screenshots.php?which=export_textfile * Browse and edit table-data using a comfortable grid http://www.heidisql.com/screenshots.php?which=data * Bulk edit tables (move to db, change engine, collation etc.) * Batch-insert ascii or binary files into tables http://www.heidisql.com/screenshots.php?which=insert_files * Write queries with customizable syntax-highlighting and code-completion http://www.heidisql.com/screenshots.php?which=query * Pretty reformat disordered SQL * Monitor and kill client-processes http://www.heidisql.com/screenshots.php?which=host_processlist * Find specific text in all tables of all databases of one server http://www.heidisql.com/screenshots.php?which=find_text_on_server * Optimize and repair tables in a batch manner http://www.heidisql.com/screenshots.php?which=maintenance * Launch a parallel mysql.exe command line window using your current connection settings * And much more http://www.heidisql.com/screenshots.php if you mind, would you give me a tutorial or anything to configure that? thank you so much On Thu, Sep 11, 2014 at 3:46 PM, Rainer Piper rainer.pi...@soho-piper.de mailto:rainer.pi...@soho-piper.de wrote: Am 11.09.2014 um 10:36 schrieb rafa alfurqan: Hi, thank you for your repplied, As you're on Ubuntu, you can begin with $ sudo apt-get install phpmyadmin i did that, so what i have to do for the configuration in asterisk so i could remote to asterisk database from phpmyadmin? Also, 10.04 is a really old Ubuntu release now, even although it is a Long Term Support one. Consider upgrading to 14.04. You can apt-get dist-upgrade straight from an LTS release to the next LTS release, without needing to go through all the intermediate releases. really appreciate for the advice, i'll do that after i could remote to asterisk database from phpmyadmin. actually i have installed freeradius-server on my ubuntu too, and i could remote the database freeradius from phpmyadmin. is it possible if same phpmyadmin could remote database from freeradius-server and asterisk (they are on same server)? thank you are you sure about allowing remote access to phpmyadmin ??? think about security first !!! I suggest HeidiSQL Client at your Home PC connecting via SSH Tunnel to your remote mySQL listening at localhost only. link to heidiSQL - http://www.heidisql.com/ -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123
Re: [asterisk-users] Asterisk with PJSIP
Hi, can you check the Linphone Extension 9002!! The port is missing! Contact: 9002/sip:9002@192.168.177.189 mailto:sip%3A9002@192.168.177.189: Avail 24.210 Regards Rainer Am 05.09.2014 um 11:55 schrieb エムディーシー太郎: Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. --https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone is busy/congested at this time (1:0/0/1)'? (1:0/0/1---num.nochan is 1.) -- 1. endpoint *CLI pjsip show endpoints Endpoint: Endpoint/CID. State. Channels. I/OAuth: AuthId/UserName... Aor: Aor MaxContact Contact: Aor/ContactUri... Status RTT(ms).. Transport: TransportId Type cos tos BindAddress.. Identify: MatchList. Channel: ChannelId.. State. Time(sec) Exten: DialedExten... CLCID: ConnectedLineCID... = Endpoint: 9001 Not in use0 of inf InAuth: auth9001/9001 Aor: 9001 10 Contact: 9001/sip:9001@192.168.177.180:16060 http://sip:9001@192.168.177.180:16060 Avail 25.048 Transport: transport-udp udp 0 0 0.0.0.0:5060 http://0.0.0.0:5060 Endpoint: 9002 Not in use0 of inf InAuth: auth9002/9002 Aor: 9002 10 * Contact: 9002/**sip:9002@192.168.177.189 mailto:sip%3A9002@192.168.177.189**Avail 24.210* Transport: transport-udp udp 0 0 0.0.0.0:5060 http://0.0.0.0:5060 -- 2. dial from 9001 to 9002 *CLI -- Executing [9002@internal:1] Dial(PJSIP/9001-, PJSIP/9002,20) in new stack -- Called PJSIP/9002 == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'PJSIP/9001-' status is 'CHANUNAVAIL' -- Thanks, MMEEGGAA -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test) -- _ -- 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] PJSIP issues with handling incoming calls
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP issues with handling incoming calls
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just
look at the cli output *NoOp( 49${gotoadr:-11} )
Am 02.09.2014 um 17:04 schrieb Rainer Piper:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
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Re: [asterisk-users] PJSIP issues with handling incoming calls
upps and delete the 49 in *Goto(49${gotoadr:-11},1) and *NoOp(
49${gotoadr:-11} )
*just look at the cli output*
Am 02.09.2014 um 17:25 schrieb Rainer Piper:
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just
look at the cli output *NoOp( 49${gotoadr:-11} )
Am 02.09.2014 um 17:04 schrieb Rainer Piper:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- 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] PJSIP issues with handling incoming calls
contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi
script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de
mailto:rainer.pi...@soho-piper.de wrote:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- 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
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] PJSIP issues with handling incoming calls
Am 02.09.2014 um 20:11 schrieb Rainer Piper: username ? -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test) -- _ -- 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] PJSIP issues with handling incoming calls
contact_user in pjsip.conf has to point to the filter or to an agi in
the extentions.conf
like:
pjsip.conf
contact_user=*blablabla
extensions.conf
**exten = blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
*
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi
script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de
mailto:rainer.pi...@soho-piper.de wrote:
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
*extensions.conf* ; i'm cutting the dialed number out of the invite
Header and goto/jump to the extensions
; incoming VOIP 9716716x SIPGATE
exten = sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
same = n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
same = n,NoOp( 49${gotoadr:-11} )
same = n,*Goto(49${gotoadr:-11},1)*
; the filter is jumping to the extensions ...
; incoming VOIP 97167160 SIPGATE - MENU
exten =
4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000PJSIP/7004PJSIP/7003PJSIP/7005,,r)
; incoming VOIP 97167161 SIPGATE
exten =
4922897167161,1,Dial(PJSIP/7000PJSIP/7001PJSIP/7003PJSIP/7004,,r)
; incoming VOIP 97167162 SIPGATE ECHO TEST
exten =
4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167163 SIPGATE
exten =
4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167164 SIPGATE
exten =
4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167165 SIPGATE
exten =
4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incncoming VOIP 97167166 Mailbox
exten =
4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167167 CONF. 1
exten =
4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
; incoming VOIP 97167168 CONF. 2
;exten =
4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
exten = 4922897167168,1,Answer
same = n,echo()
same = n,Hangup()
; incoming VOIP 97167169 FAX
;exten =
4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.dePJSIP/7000,,r)
Regards
Rainer
Am 02.09.2014 um 15:08 schrieb Joshua Colp:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused
how that... works...
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- 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
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Copying menuselect options
I compile everything and then disable the unwanted modules in modules.conf like: modules.conf: ; ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes preload = res_odbc.so preload = res_config_odbc.so ;noload = res_odbc.so ;noload = res_config_odbc.so noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so load = res_musiconhold.so noload = chan_alsa.so noload = chan_oss.so noload = chan_console.so noload = chan_sccp.so noload = chan_skinny.so noload = chan_mgcp.so noload = pbx_dundi.so noload = chan_iax2.so noload = chan_unistim.so noload = res_corosync.so noload = res_xmpp.so noload = res_ari.so noload = pbx_ael.so noload = chan_sip.so ;noload = chan_pjsip.so noload = res_config_ldap.so noload = chan_motif.so noload = res_fax.so noload = res_fax_spandsp.so noload = res_config_mysql.so noload = bridge_native_rtp.so noload = func_odbc.so noload = res_ari_applications.so noload = res_ari_bridges.so noload = res_ari_device_states.so noload = res_ari_events.so noload = res_ari_model.so noload = res_ari_recordings.so noload = res_ari_sounds.so noload = res_ari_asterisk.so noload = res_ari_channels.so noload = res_ari_endpoints.so noload = res_ari_mailboxes.so noload = res_ari_playbacks.so noload = res_ari.so noload = cel_custom.so noload = cel_manager.so noload = cel_odbc.so noload = cel_pgsql.so noload = cel_radius.so noload = cel_sqlite3_custom.so noload = cel_tds.so noload = cdr_pgsql.so noload = res_config_pgsql.so noload = app_morsecode.so noload = res_phoneprov.so noload = app_ices.so noload = app_macro.so noload = app_festival.so noload = app_page.so noload = app_alarmreceiver.so Am 15.08.2014 um 11:32 schrieb Thorsten Göllner: Am 14.08.2014 17:22, schrieb Mitch Claborn: Is it possible (and advisable) to copy menuselect options from Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only file to copy? I am not sure - but I would'nt do that. Make a hardcopy from your console and transcribe the settings to your new installation. It yould take you not more than 10 minutes. -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test) -- _ -- 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] quickstart
Am 17.06.2014 09:04, schrieb thufir: I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir Hi ... this script will get you up and running on a debian7 distribution. code |#!/bin/sh| |apt-get update apt-get upgrade -y| |asteriskversion=asterisk-12.3.2| | apt-get install -y linux-headers-`uname -r` apt-get install -y build-essential apt-get install -y wget apt-get install -y libssl-dev apt-get install -y libncurses5-dev apt-get install -y libnewt-dev apt-get install -y libxml2-dev apt-get install -y libsqlite3-dev apt-get install -y libjansson-dev apt-get install -y git ln -s /usr/src/linux-headers-`uname -r` /usr/src/linux cd /usr/src ## pjsip installieren git clone https://github.com/asterisk/pjproject pjproject cd /usr/src/pjproject ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr ## um IPv6 Support in pjsip einzuschalten, muss das CFLAGS='-DPJ_HAS_IPV6=1' angegeben werden # IPV6 is turned off at default ! #./configure CFLAGS='-DPJ_HAS_IPV6=1' --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr # make dep make make install ldconfig ### check inst. # ldconfig -p | grep libpj ## System vorbereiten ## download Asterisk if [ ! -f /usr/src/$asteriskversion.tar.gz ] ; then wget http://downloads.asterisk.org/pub/telephony/asterisk/$asteriskversion.tar.gz fi if [ ! -d /usr/src/$asteriskversion ] ; then tar xvzf $asteriskversion.tar.gz fi ## erforderliche libs installieren /usr/src/$asteriskversion/contrib/scripts/install_prereq install ## optional /usr/src/$asteriskversion/contrib/scripts/get_mp3_source.sh /usr/src/$asteriskversion/contrib/scripts/get_ilbc_source.sh gcc -O2 /usr/src/$asteriskversion/contrib/utils/rawplayer.c -o /usr/bin/rawplayer ## asterisk installieren cd /usr/src/$asteriskversion ./configure make menuconfig make make install make samples make config make install-logrotate | |/code | -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] quickstart
Am 17.06.2014 17:36, schrieb thufir: On Tue, 17 Jun 2014 12:14:05 +0200, Rainer Piper wrote: git clone https://github.com/asterisk/pjproject pjproject At the very least, thank you for pjsip. I'm not sure what it is yet, but seems intriguing :) I'm on Ubunutu 14.04, but will look over your script and adapt it. !!! take a look at the install_prereq script. !!! you have to install same dependency libs before you compile asterisk !!! and install_prereq just supports ... debian, redhat and OpenBSD # The distributions we do support: if [ -r /etc/debian_version ]; then handle_debian elif [ -r /etc/redhat-release ]; then handle_rh elif [ $OS = 'OpenBSD' ]; then handle_obsd fi -Thufir -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 callto:004922897167161 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Hi Joshua, I'll give it a try at ttps://issues.asterisk.org/jira and I hope my English is good enough to explain the problem. ;-) wow ... early bird it is 03:36 (PDT) in the morning at your place Thanks! Rainer Am 07.05.2014 12:36, schrieb Joshua Colp: Rainer Piper wrote: perhaps a silly question ... if a channel switches from simple_bridge to native_bridge ... is the channel switching to direct_media between the endpoints ? if so, why doesn't turn direct_media = no and disable_direct_media_on_nat = yes switching to native_bridge off ? The bridge_native_rtp module can actually native bridge in two ways: 1. Media directly between both sides 2. Media within the RTP stack Even with NAT #2 can still operate fine as media still goes through Asterisk, just not as much. As for your issue I would suggest you get a complete console log output with debug and create an issue[1] as this sounds like a bug. Cheers, [1] https://issues.asterisk.org/jira -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
and I get ready for launch in germany at 13:15 ;-) Am 07.05.2014 13:09, schrieb Joshua Colp: Rainer Piper wrote: Hi Joshua, I'll give it a try at ttps://issues.asterisk.org/jira and I hope my English is good enough to explain the problem. ;-) wow ... early bird it is 03:36 (PDT) in the morning at your place The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada though where it is 8:09AM. Not t early. -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
upps ... off topic and typo lunch not launch ;-) Am 07.05.2014 13:14, schrieb Rainer Piper: and I get ready for launch in germany at 13:15 ;-) Am 07.05.2014 13:09, schrieb Joshua Colp: Rainer Piper wrote: Hi Joshua, I'll give it a try at ttps://issues.asterisk.org/jira and I hope my English is good enough to explain the problem. ;-) wow ... early bird it is 03:36 (PDT) in the morning at your place The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada though where it is 8:09AM. Not t early. -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] Video with asterisk12 and pjsip
Hi, I tried to turn on Video and get the following cli-WARNING output -- Executing [8600@outgoing-kamailio:1] Answer(PJSIP/7000-, ) in new stack 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 -- Executing [8600@outgoing-kamailio:2] ConfBridge(PJSIP/7000-, 8600) in new stack -- PJSIP/7000- Playing 'conf-onlyperson.g722' (language 'de') -- PJSIP/7000- Playing 'confbridge-join.g722' (language 'de') -- CBAnn/8600-;1 Playing 'confbridge-join.slin' (language 'en') -- Channel CBAnn/8600-;2 joined 'softmix' base-bridge 52997aa1-eb00-481c-8c56-e26d78d01515 -- Channel CBAnn/8600-;2 left 'softmix' base-bridge 52997aa1-eb00-481c-8c56-e26d78d01515 -- Started music on hold, class 'default', on channel 'PJSIP/7000-' -- Channel PJSIP/7000- joined 'softmix' base-bridge 52997aa1-eb00-481c-8c56-e26d78d01515 [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 0x7f46f4187280 -- Probation passed - setting RTP source address to 192.168.8.203:31384 Endpoint 7000 is a Grandstream GXV3175 with Video the pjsip.conf for exten 7000 is [7000] type=endpoint context=outgoing-kamailio disallow=all allow=g722,alaw,ulaw,h264,h263p,h263,h261 transport=transport-udp auth=auth7000 aors=7000 direct_media=no disable_direct_media_on_nat=yes do I have to turn on the Video Support somewhere else ? -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY server*CLI core show channel PJSIP/7000-0001 -- General -- Name: PJSIP/7000-0001 Type: PJSIP UniqueID: 1399382022.1 LinkedID: 1399382022.0 Caller ID: 7000 Caller ID Name: (N/A) Connected Line ID: 7001 Connected Line ID Name: 7001 Eff. Connected Line ID: 7001 Eff. Connected Line ID Name: 7001 DNID Digits: (N/A) Language: de State: Up (6) NativeFormats: (alaw) WriteFormat: g722 ReadFormat: g722 WriteTranscode: Yes (g722)-(slin)-(alaw) ReadTranscode: Yes (alaw)-(slin)-(g722) Time to Hangup: 0 Elapsed Time: 0h3m24s Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148 -- PBX -- Context: outgoing-kamailio Extension:pjsi Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppDial Data: (Outgoing Line) Call Identifer: [C-] Variables: BRIDGEPEER=PJSIP/7001- DIALEDPEERNUMBER=7000 CDR Variables: level 1: calledsubaddr= level 1: callingsubaddr= level 1: dnid= level 1: clid= 700 level 1: src=7000 level 1: dcontext=outgoin level 1: channel=PJSIP/7 level 1: lastapp=AppDial level 1: lastdata=(Outgoi level 1: start=1399382 level 1: answer=1399382 level 1: end=1399382 level 1: duration=1 level 1: billsec=0 level 1: disposition=8 level 1: amaflags=3 level 1: uniqueid=1399382 level 1: linkedid=1399382 level 1: sequence=1 server*CLI core show channel PJSIP/7001- -- General -- Name: PJSIP/7001- Type: PJSIP UniqueID: 1399382022.0 LinkedID: 1399382022.0 Caller ID: 7001 Caller ID Name: 7001 Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: de State: Up (6) NativeFormats: (g722) WriteFormat: g722 ReadFormat: g722 WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h3m51s Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148 -- PBX -- Context: outgoing-kamailio Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: Dial Data: PJSIP/7000 Call Identifer: [C-] Variables: BRIDGEPEER=PJSIP/7000-0001 DIALEDPEERNUMBER=7000 DIALEDPEERNAME=PJSIP/7000-0001 DIALSTATUS=ANSWER DIALEDTIME= ANSWEREDTIME= CDR Variables: level 1: calledsubaddr= level 1: callingsubaddr= level 1: dnid= level 1: clid=7001 level 1: src=7001 level 1: dst=7000 level 1: dcontext=outgoin level 1: channel=PJSIP/7 level 1: dstchannel=PJSIP/7 level 1: lastapp=Dial level 1: lastdata=PJSIP/7 level 1: start=1399382 level 1: answer=1399382 level 1: end=0.0 level 1: duration=230 level 1: billsec=228 level 1: disposition=8 level 1: amaflags=3 level 1: uniqueid=1399382 level 1: linkedid=1399382 level 1: sequence=0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
that's funny I recompiled asterisk without bridge_native_rtp.so to force asterisk to go to simple_bridge and not to native_bridge... !!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu Am 07.05.2014 07:11, schrieb Rainer Piper: PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de NOC +49 228 97167161 - sip.soho-piper.de NOC +882 990111550 via e164.org International Network NOC +49 2247 9064188 - sip.tefonix.de - D293 NOC +882 990045450 via e164.org International Network -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
perhaps a silly question ... if a channel switches from simple_bridge to native_bridge ... is the channel switching to direct_media between the endpoints ? if so, why doesn't turn direct_media = no and disable_direct_media_on_nat = yes switching to native_bridge off ? my pjsip.conf endpoint 7000 and 7001 [7000] type=endpoint context=outgoing disallow=all allow=alaw,ulaw,g722 transport=transport-udp auth=auth7000 aors=7000 direct_media = no disable_direct_media_on_nat = yes [auth7000] type=auth auth_type=userpass password=x username=7000 [7000] type=aor max_contacts=10 qualify_frequency=60 [7001] type=endpoint context=outgoing disallow=all allow=g722,alaw,ulaw transport=transport-udp auth=auth7001 aors=7001 direct_media = no disable_direct_media_on_nat = yes [auth7001] type=auth auth_type=userpass password=x username=7001 [7001] type=aor max_contacts=10 qualify_frequency=60 Am 07.05.2014 07:35, schrieb Rainer Piper: that's funny I recompiled asterisk without bridge_native_rtp.so to force asterisk to go to simple_bridge and not to native_bridge... !!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu Am 07.05.2014 07:11, schrieb Rainer Piper: PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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
