[asterisk-users] question about zapata.conf
hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command Thanks and regards -- _ -- 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] Need help about round-robin
thanks a lot i will test and i will update you as soon as i have any
problem
2013/3/22 Asghar Mohammad asghar...@gmail.com
your dialplan nothing to do with bandwidth it dial out to digium card what
ever come in.
1.
if your providers calls come in via digium card and you want send out
using sip or any other tech. then use context defined in group 1 for
provider 1 and context defined in group 2 for provider 2.
2.
if your providers come in using sip just give him deferent ips, provider 1
send to wimax ip and provider to FH.
or explain if you are using other scenario.
On Fri, Mar 22, 2013 at 7:14 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
yes i want to use the burden-sharing between Wimax and FH using a diguim
cards
2013/3/22 Asghar Mohammad asghar...@gmail.com
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about burden-sharing between Wimax and FH?
On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
ok thank you so much i use dial(zap/r2) instead of g2 and it works
without problem
now my question i have 2 providers i use g1 for the first and g2 for
the second
if i understand i must use r1 instead of g1 for the first provider and
r2 instead of g2 for the second provider in order to use the burden-sharing
between Wimax and FH
thanks and regards
2013/3/21 Asghar Mohammad asghar...@gmail.com
hi,
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r in Dial.
you can use r for Ascending and R for Descending order
On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten =
_0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();
thanks and regards.
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i
use the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.
Note that the direction of the 'slash' is significant as is the
leading slash.
--
Thanks in advance,
--**--**
-
Steve Edwards sedwa...@sedwards.com Voice:
+1-760-468-3867 PST
Newline Fax:
+1-760-731-3000
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Re: [asterisk-users] question about zapata.conf
it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- 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] question about zapata.conf
i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Optimizing Asterisk Environment
Hello Guys, Thank you so much for your response. We reran the sipp test: ./sipp -sf uac_pcap.xml -s 1001 vancouver.example.com -l 250 -trace_err -mp 3 -d 1 The scenario is the standard contact asterisk play some rtp media. On the asterisk, the echo test was used for the extension. This simulating a two way audio test. With ulimit set ulimit -n 65535, and while the test was running: # top PIDPR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 16056 20 0 67568 25m 5812 S36 0.7 0:36.90asterisk #iftop (nice tool by the way :) vancouver.test.com = 192.168.2.100 1.75Mb1.75Mb 1.75Mb = 1.69Mb 1.70Mb 1.70Mb # free -m total used free sharedbuffers cached Mem: 3735518 3217 0 30438 -/+ buffers/cache: 48 3686 Swap: 2047 0 2047 # uptime 10:55:09 up 2 days, 1:45, 1 user, load average: 0.44, 0.46, 0.23 #ifconfig UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:13222747 errors:0 dropped:0 overruns:1 frame:1 TX packets:62311814 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 # vmstat procs ---memory-- ---swap-- -io -system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 0 0 0 3293204 31824 45042400 1 1 931 1 1 98 0 # dmesg | grep -i duplex [ 14.622293] e100 :00:02.0: eth3: NIC Link is Up 100 Mbps Full Duplex We are running this on a test server (x330) just to help us with the dimensioning process for now. The important results from SIPP: Call-rate(length) Port Total-time Total-calls Remote-host 10.0(1 ms)/1.000s 5060 654.01 s 6450 192.168.2.10:5060(UDP) 0 calls (limit 250)Peak was 91 calls, after 9 s Elapsed Time |00:10:54:030 Call Rate |9.862 cps Successful call |0 | 6450 Failed call|0 |0 Is it safe to say that our test router is a lemon? Not sure if that's the bottleneck at this moment. Since only 36% of CPU is being utilized, and only 0.7% of memory. Are there any setting I should double check to run asterisk in full capacity. Thank you so much for your help, Nick. On 3/24/13, Steve Edwards asterisk@sedwards.com wrote: On Sat, Mar 23, 2013 at 09:33:38AM -0400, Nick Khamis wrote: We are getting some rather poor results (relative) with our Asterisk setup. On Sun, 24 Mar 2013, Tzafrir Cohen wrote: Run the system in full capacity and provide us some data. For starters: free -m uptime vmstat ethtool - make sure the interfaces are set correctly - look for 'Speed: 1000Mb/s' and 'Duplex: Full' ifconfig - look at the error counters iftop - how many bits are you pushing in each direction I've got a 7 year old Xeon box with 2GB running Asterisk 1.2 that handles 300 channels just fine. I suspect a modern box with a modern Asterisk could do that in 'sleep mode.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Optimizing Asterisk Environment
On the network side, Joshua had mentioned some network based encoding solutions. We would be sold on this. What are some of the routers out there that provide this capability with descent throughput? We were considering Cisco 3745 with a NM-HDV2 which transcoding from G.711 to G.729 handles 96 sessions. Not sure if this was the best bang for our buck? N. On 3/25/13, Nick Khamis sym...@gmail.com wrote: Hello Guys, Thank you so much for your response. We reran the sipp test: ./sipp -sf uac_pcap.xml -s 1001 vancouver.example.com -l 250 -trace_err -mp 3 -d 1 The scenario is the standard contact asterisk play some rtp media. On the asterisk, the echo test was used for the extension. This simulating a two way audio test. With ulimit set ulimit -n 65535, and while the test was running: # top PIDPR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 16056 20 0 67568 25m 5812 S36 0.7 0:36.90asterisk #iftop (nice tool by the way :) vancouver.test.com = 192.168.2.100 1.75Mb1.75Mb 1.75Mb = 1.69Mb 1.70Mb 1.70Mb # free -m total used free sharedbuffers cached Mem: 3735518 3217 0 30438 -/+ buffers/cache: 48 3686 Swap: 2047 0 2047 # uptime 10:55:09 up 2 days, 1:45, 1 user, load average: 0.44, 0.46, 0.23 #ifconfig UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:13222747 errors:0 dropped:0 overruns:1 frame:1 TX packets:62311814 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 # vmstat procs ---memory-- ---swap-- -io -system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 0 0 0 3293204 31824 45042400 1 1 931 1 1 98 0 # dmesg | grep -i duplex [ 14.622293] e100 :00:02.0: eth3: NIC Link is Up 100 Mbps Full Duplex We are running this on a test server (x330) just to help us with the dimensioning process for now. The important results from SIPP: Call-rate(length) Port Total-time Total-calls Remote-host 10.0(1 ms)/1.000s 5060 654.01 s 6450 192.168.2.10:5060(UDP) 0 calls (limit 250)Peak was 91 calls, after 9 s Elapsed Time |00:10:54:030 Call Rate |9.862 cps Successful call |0 | 6450 Failed call|0 |0 Is it safe to say that our test router is a lemon? Not sure if that's the bottleneck at this moment. Since only 36% of CPU is being utilized, and only 0.7% of memory. Are there any setting I should double check to run asterisk in full capacity. Thank you so much for your help, Nick. On 3/24/13, Steve Edwards asterisk@sedwards.com wrote: On Sat, Mar 23, 2013 at 09:33:38AM -0400, Nick Khamis wrote: We are getting some rather poor results (relative) with our Asterisk setup. On Sun, 24 Mar 2013, Tzafrir Cohen wrote: Run the system in full capacity and provide us some data. For starters: free -m uptime vmstat ethtool - make sure the interfaces are set correctly - look for 'Speed: 1000Mb/s' and 'Duplex: Full' ifconfig - look at the error counters iftop - how many bits are you pushing in each direction I've got a 7 year old Xeon box with 2GB running Asterisk 1.2 that handles 300 channels just fine. I suspect a modern box with a modern Asterisk could do that in 'sleep mode.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] question about zapata.conf
hi, migrating from zaptel to dahdi HAS an impact... new config files, new options and a new channeldriver that has to be used in your dialplan ... you would have to select the DAHDI channel instead of your ZAP channel when dialing... if you´re to afraid to do it... then leave it as it is and follow the ntars-maxime (never touch a running system)... regards, yves Am 25.03.2013 16:15, schrieb Salaheddine Elharit: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com mailto:ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de mailto:yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] question about zapata.conf
ok thank you so much for your help and support 2013/3/25 Yves A. yves...@gmx.de hi, migrating from zaptel to dahdi HAS an impact... new config files, new options and a new channeldriver that has to be used in your dialplan ... you would have to select the DAHDI channel instead of your ZAP channel when dialing... if you´re to afraid to do it... then leave it as it is and follow the ntars-maxime (never touch a running system)... regards, yves Am 25.03.2013 16:15, schrieb Salaheddine Elharit: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using type=friend a mistake?
Hello Everyone, Just looking to secure our * box, and stumbled on the following This advice may run counter to the majority of documentation, sample files and examples shown on the voip-info.org site and on Asterisk forums, but you’ll have to take my word for it – using “type=friend” is a big mistake! It will make your Asterisk server much more vulnerable because “type=friend” actually causes two objects to be created – a SIP peer and a SIP user. This gives the potential hacker two entrance doors into your PBX, one of which has comparatively weak security. The problem is that a “user” is allowed to connect from any remote IP address, not just the address specified in the host parameter. Even if you want to allow connections from any address, it is much better to use “host=dynamic” than to use “type=friend”., http://kb.smartvox.co.uk/asterisk/secure-asterisk-pbx-part-2/ Is this true? Before I update all my type to peer, what are some of the things we needs to keep in mind when using friend vs. peer from a security standpoint? Thanks in Advance, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11, hangup-handlers, Local channels and channel originate
Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when the second channel hangs up. At the moment, I'm issuing a couple of channel originate Local/1@mycontext1 extension 123456789@mycontext2 commands. I'm observing that as I'm using expressions such as Local/1@mycontext1, a Local ZOMBIE channel is hanged before the second channel stops ringing. When the second channel itself ends, my handler is not run anymore. What would you suggest me to do ? Should I delay my Set(CHANNEL(hangup_handler_wipe)= ...) statement till both channels are bridged together ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11, hangup-handlers, Local channels and channel originate
On 03/25/2013 05:17 PM, Olivier wrote: Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when the second channel hangs up. At the moment, I'm issuing a couple of channel originate Local/1@mycontext1 extension 123456789@mycontext2 commands. I'm observing that as I'm using expressions such as Local/1@mycontext1, a Local ZOMBIE channel is hanged before the second channel stops ringing. When the second channel itself ends, my handler is not run anymore. What would you suggest me to do ? Should I delay my Set(CHANNEL(hangup_handler_wipe)= ...) statement till both channels are bridged together ? It is hard to say without seeing the dialplan that you're using. Most likely, the hangup handler has been attached to one half of the Local channel as opposed to the channel you want it attached to. Can you include the full dialplan that you're using? Matt -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11, hangup-handlers, Local channels and channel originate
2013/3/25 Matthew Jordan mjor...@digium.com On 03/25/2013 05:17 PM, Olivier wrote: Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when the second channel hangs up. At the moment, I'm issuing a couple of channel originate Local/1@mycontext1 extension 123456789@mycontext2 commands. I'm observing that as I'm using expressions such as Local/1@mycontext1, a Local ZOMBIE channel is hanged before the second channel stops ringing. When the second channel itself ends, my handler is not run anymore. What would you suggest me to do ? Should I delay my Set(CHANNEL(hangup_handler_wipe)= ...) statement till both channels are bridged together ? It is hard to say without seeing the dialplan that you're using. Most likely, the hangup handler has been attached to one half of the Local channel as opposed to the channel you want it attached to. Can you include the full dialplan that you're using? Yes, of course. I'll simplify it and post it here ASAP. Matt -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11, hangup-handlers, Local channels and channel originate
2013/3/25 Olivier oza_4...@yahoo.fr
2013/3/25 Matthew Jordan mjor...@digium.com
On 03/25/2013 05:17 PM, Olivier wrote:
Hello,
I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup.
My plan is to use this handler to update my CDRs with values such as
Asterish and Tech cause (see function HANGUP_CAUSE).
I want to have my custom hangup-handler be run only once and when the
second channel hangs up.
At the moment, I'm issuing a couple of channel originate
Local/1@mycontext1 extension 123456789@mycontext2 commands.
I'm observing that as I'm using expressions such as Local/1@mycontext1,
a Local ZOMBIE channel is hanged before the second channel stops
ringing.
When the second channel itself ends, my handler is not run anymore.
What would you suggest me to do ?
Should I delay my Set(CHANNEL(hangup_handler_wipe)= ...) statement till
both channels are bridged together ?
It is hard to say without seeing the dialplan that you're using. Most
likely, the hangup handler has been attached to one half of the Local
channel as opposed to the channel you want it attached to. Can you
include the full dialplan that you're using?
Yes, of course.
I'll simplify it and post it here ASAP.
Here it is:
[hangup-handler]
exten = s,1,Verbose(0,Entering context ${CONTEXT} in channel ${CHANNEL}
with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)})
[to-foobar]
exten = _X.,1,Verbose(0,Entering context ${CONTEXT} with EXTEN and CID set
to ${EXTEN} and ${CALLERID(num)})
same = n, Set(CHANNEL(hangup_handler_push)=hangup-handler,s,1)
same = n, Dial(SIP/foobar/${EXTEN})
same = n, Hangup()
[from-foobar]
exten = _X.,1,Verbose(0,Entering context ${CONTEXT} with EXTEN and CID set
to ${EXTEN} and ${CALLERID(num)})
same = n, Dial(SIP/foobar/${EXTEN})
same = n, Hangup()
The command I used is :
channel originate Local/7005@from-foobar extension 7003@to-foobar
Console prints:
Entering context from-foobar with EXTEN and CID set to 7005 and
Entering context to-foobar with EXTEN and CID set to 7003 and
Entering context hangup-handler in channel
Local/7005@from-foobar-0008;1ZOMBIE
with EXTEN and CID set to s and
The first line is printed at soon as Enter key is pressed.
The second and third lines are printed when originating channel answers
(here extension SIP/foobar/7005)
Matt
--
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
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Re: [asterisk-users] Asterisk 11, hangup-handlers, Local channels and channel originate
On 03/25/2013 05:17 PM, Olivier wrote:
Hello,
I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup.
My plan is to use this handler to update my CDRs with values such
as
Asterish and Tech cause (see function HANGUP_CAUSE).
I want to have my custom hangup-handler be run only once and when
the
second channel hangs up.
At the moment, I'm issuing a couple of channel originate
Local/1@mycontext1 extension 123456789@mycontext2 commands.
I'm observing that as I'm using expressions such as
Local/1@mycontext1,
a Local ZOMBIE channel is hanged before the second channel stops
ringing.
When the second channel itself ends, my handler is not run anymore.
What would you suggest me to do ?
Should I delay my Set(CHANNEL(hangup_handler_wipe)= ...) statement
till
both channels are bridged together ?
It is hard to say without seeing the dialplan that you're using. Most
likely, the hangup handler has been attached to one half of the Local
channel as opposed to the channel you want it attached to. Can you
include the full dialplan that you're using?
Yes, of course.
I'll simplify it and post it here ASAP.
Here it is:
[hangup-handler]
exten = s,1,Verbose(0,Entering context ${CONTEXT} in channel
${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)})
[to-foobar]
exten = _X.,1,Verbose(0,Entering context ${CONTEXT} with EXTEN and
CID set to ${EXTEN} and ${CALLERID(num)})
same = n, Set(CHANNEL(hangup_handler_push)=hangup-handler,s,1)
same = n, Dial(SIP/foobar/${EXTEN})
same = n, Hangup()
[from-foobar]
exten = _X.,1,Verbose(0,Entering context ${CONTEXT} with EXTEN and
CID set to ${EXTEN} and ${CALLERID(num)})
same = n, Dial(SIP/foobar/${EXTEN})
same = n, Hangup()
The command I used is :
channel originate Local/7005@from-foobar extension 7003@to-foobar
Console prints:
Entering context from-foobar with EXTEN and CID set to 7005 and
Entering context to-foobar with EXTEN and CID set to 7003 and
Entering context hangup-handler in channel
Local/7005@from-foobar-0008;1ZOMBIE with EXTEN and CID set to
s and
The first line is printed at soon as Enter key is pressed.
The second and third lines are printed when originating channel
answers (here extension SIP/foobar/7005)
The originate creates a chain of channels like so:
SIP/foobar/7005 -- Local/7005@from-foobar;1 -- Local/7005@from-foobar;2 --
SIP/foobar/7003
You put the hangup handler on the Local/7005@from-foobar;2 channel. When
the local channel optimizes itself out, the hangup handler is run on the
hanging up local channel.
What you need to do is use a pre-dial handler to put the hangup handler on
the SIP/foobar/7003 channel.
Richard
[1] https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers
[2] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
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