Re: [asterisk-users] (no subject)
If you're using a redhat based distro, have you checked SELinux? Try disabling (will require a server reboot) Regards Ish On 3 September 2014 20:41, Steve Edwards asterisk@sedwards.com wrote: For future reference, a well chosen subject will yield more relevant replies. Better bait == better fish. -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] opus 11.12.0
hi, any plans update patch for 11.12.0? |https://github.com/meetecho/asterisk-opus https://github.com/netaskd/asterisk-opus/ | patching file build_tools/menuselect-deps.in patching file channels/chan_sip.c Hunk #1 succeeded at 7659 (offset -98 lines). Hunk #2 succeeded at 11011 (offset -34 lines). Hunk #3 succeeded at 11050 (offset -34 lines). Hunk #4 succeeded at 7 with fuzz 1 (offset -34 lines). Hunk #5 FAILED at 12722. 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file codecs/codec_opus.c patching file codecs/ex_opus.h patching file configure.ac Hunk #2 succeeded at 2150 (offset 31 lines). patching file formats/format_vp8.c patching file include/asterisk/format.h patching file main/channel.c patching file main/format.c Hunk #6 succeeded at 1098 (offset 12 lines). patching file main/frame.c patching file main/rtp_engine.c Hunk #1 succeeded at 2326 (offset 37 lines). Hunk #2 succeeded at 2370 (offset 37 lines). patching file makeopts.in patching file res/res_rtp_asterisk.c Hunk #1 succeeded at 95 with fuzz 1 (offset 4 lines). Hunk #2 FAILED at 349. Hunk #3 succeeded at 3011 (offset 394 lines). Hunk #4 succeeded at 3097 (offset 394 lines). 1 out of 4 hunks FAILED -- saving rejects to file res/res_rtp_asterisk.c.rej thanks -- --- Marek Cervenka === -- _ -- 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 secure fine tune - stop attack
Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. Thanks in advance, -Motty -- _ -- 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 secure fine tune - stop attack
On 04-09-14 16:44, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306 http://213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. Have a look at Fail2ban: http://www.fail2ban.org/wiki/index.php/Main_Page HTH, Patrick -- _ -- 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 secure fine tune - stop attack
Am 04.09.2014 16:44, schrieb motty cruz: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306 http://213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. You should not invest time in blocking single IPs. Take a look at fail2ban. http://www.fail2ban.org/wiki/index.php/Asterisk -Thorsten- -- _ -- 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 secure fine tune - stop attack
Thanks, looks like fail2ban is the way to go, I would prefer a different alternatives if there is one. I tried deny=IP/netmask but did not work for me, in sip.conf. seems like fail2ban is what you all are using, so I will give it a try. Thanks, On Thu, Sep 4, 2014 at 7:58 AM, Thorsten Göllner t...@ovm-group.com wrote: Am 04.09.2014 16:44, schrieb motty cruz: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. You should not invest time in blocking single IPs. Take a look at fail2ban. http://www.fail2ban.org/wiki/index.php/Asterisk -Thorsten- -- _ -- 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] Unable to connect to remote asterisk
solved, permissions problem. Asterisks run with user asterisk at default, I changed to asteriskpbx as the book says ;) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Azzopardi Sent: 03 September 2014 20:57 To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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 secure fine tune - stop attack
On Thursday 04 Sep 2014, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? Instead of blocking unwanted IPs, you should be permitting only wanted IPs. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 secure fine tune - stop attack
Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. Thanks for your suggestions, I will try fail2ban. I don't know how complicated is to implement that on production server. Thanks, -Motty On Thu, Sep 4, 2014 at 8:19 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 04 Sep 2014, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? Instead of blocking unwanted IPs, you should be permitting only wanted IPs. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 secure fine tune - stop attack
dont forgot to put your trusted IPs into ignoreip list while configuring fail2ban its very important when a customer (may be 100+ extns) are behind NAT and only present single public IP RgdsHash Date: Thu, 4 Sep 2014 08:42:11 -0700 From: motty.c...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk secure fine tune - stop attack Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. Thanks for your suggestions, I will try fail2ban. I don't know how complicated is to implement that on production server. Thanks, -Motty On Thu, Sep 4, 2014 at 8:19 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 04 Sep 2014, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? Instead of blocking unwanted IPs, you should be permitting only wanted IPs. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk secure fine tune - stop attack
If we don't need to allow access from outside the USA we block access from all non-ARIN IP addresses by using iptables. This takes care of at least 80% of attacks. I enabled guest access and pointed all guest calls to an IVR which auto disconnects the call after a while (2 min seems good) if there is no response. That took care of most of the remaining attacks. I'm considering enabling auto create peer and routing calls to the same IVR as above. We also use fail2ban, but mostly for non-SIP attacks. Before enabling any guest access be ABSOLUTELY SURE you know how to do it without causing security issues. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hashmat Khan Sent: Thursday, September 04, 2014 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk secure fine tune - stop attack dont forgot to put your trusted IPs into ignoreip list while configuring fail2ban its very important when a customer (may be 100+ extns) are behind NAT and only present single public IP Rgds Hash Date: Thu, 4 Sep 2014 08:42:11 -0700 From: motty.c...@gmail.commailto:motty.c...@gmail.com To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk secure fine tune - stop attack Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. Thanks for your suggestions, I will try fail2ban. I don't know how complicated is to implement that on production server. Thanks, -Motty On Thu, Sep 4, 2014 at 8:19 AM, A J Stiles asterisk_l...@earthshod.co.ukmailto:asterisk_l...@earthshod.co.uk wrote: On Thursday 04 Sep 2014, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? Instead of blocking unwanted IPs, you should be permitting only wanted IPs. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Failover / modifying response time
I was able to get a packet trace of this event Time 312.353549 - INVITE to primary 313.222303 - INVITE to primary ( suspected resend of frame ) 314.289215 - INVITE to backup 315.397120 - INVITE to backup ( suspected resend of frame ) So is primary just too slow to answer ? I am not seeing anything in the logs on primary. On Wed, Sep 3, 2014 at 2:39 PM, Stephen More stephen.m...@gmail.com wrote: I have two real time asterisk boxes configured to accept incoming or outgoing calls at any time. All the users are configured to send their calls to primary. If primary is down calls will go to backup. All incoming SIP calls should be sent to primary. If primary is down incoming calls will go to backup. 99.9% of the time it works as designed. Every once in a while my SIP provider will send a sip call to our backup. primary is working fine. When asked they responded with it's possible primary didn't respond fast enough so backup grabbed it. You should lower the response time on primary and raise it on backup What setting is he referring to ? Can I tweak a setting to make our backup server respond slower so that the primary answers the call ? -Thanks -- _ -- 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 secure fine tune - stop attack
You can also take a look at SecAst (www.generationd.com).The free version is a drop-in replacement for fail2ban but also add a lot more intelligence (and no need to update regex's etc). There's also geographic IP fencing so you can block attacks by country / region / city etc., only allow access by geography, etc. And a whole lot more (including detection of breached but valid credentials to halt ongoing fraud, etc) -=M=- The opinions above are my own, and don't necessarily represent those of my employer. Since I'm employed by Generation D however you can bet that I have a serious bias :) From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Eric Wieling ewiel...@nyigc.com Sent: Thursday, September 4, 2014 11:58 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk secure fine tune - stop attack If we don't need to allow access from outside the USA we block access from all non-ARIN IP addresses by using iptables. This takes care of at least 80% of attacks. I enabled guest access and pointed all guest calls to an IVR which auto disconnects the call after a while (2 min seems good) if there is no response. That took care of most of the remaining attacks. I'm considering enabling auto create peer and routing calls to the same IVR as above. We also use fail2ban, but mostly for non-SIP attacks. Before enabling any guest access be ABSOLUTELY SURE you know how to do it without causing security issues. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hashmat Khan Sent: Thursday, September 04, 2014 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk secure fine tune - stop attack dont forgot to put your trusted IPs into ignoreip list while configuring fail2ban its very important when a customer (may be 100+ extns) are behind NAT and only present single public IP Rgds Hash Date: Thu, 4 Sep 2014 08:42:11 -0700 From: motty.c...@gmail.commailto:motty.c...@gmail.com To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk secure fine tune - stop attack Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. Thanks for your suggestions, I will try fail2ban. I don't know how complicated is to implement that on production server. Thanks, -Motty On Thu, Sep 4, 2014 at 8:19 AM, A J Stiles asterisk_l...@earthshod.co.ukmailto:asterisk_l...@earthshod.co.uk wrote: On Thursday 04 Sep 2014, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? Instead of blocking unwanted IPs, you should be permitting only wanted IPs. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://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.comhttp://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 secure fine tune - stop attack
Please don't top post. On Thu, 4 Sep 2014, motty cruz wrote: Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. Do your few extensions travel to China, Russia, Iran, Iraq, North Korea, etc? (Sorry if I stepped on anybody's toes.) If you configure iptables to drop all and then only allow the few IP address ranges you really need, 90% of the problem is solved. Then use fail2ban to manage the remaining anklebitters. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk secure fine tune - stop attack
On Thursday 04 Sep 2014, motty cruz wrote: Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. If you know what ISPs they are using, then you can allow just those ISPs' address ranges. That will slow things down, by requiring an attacker to be using the same ISP as a legitimate user. Thanks for your suggestions, I will try fail2ban. I don't know how complicated is to implement that on production server. It's fairly easy -- but note that physical access to the server's console is highly desirable, lest you accidentally block yourself out from using ssh (not a mistake you want to make too many times). -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 secure fine tune - stop attack
On 4/9/14 4:58 pm, Eric Wieling wrote: If we don't need to allow access from outside the USA we block access from all non-ARIN IP addresses by using iptables. This takes care of at least 80% of attacks. Likewise here (though RIPE rather than ARIN, since we're the other side of the pond). You can also take it a bit further: if, for example, you know what ISP(s) your dynamic clients are using, you can limit connections to the IP ranges those ISP(s) use - look up their ranges on he.net's BGP looking glass if you need to find out what ranges they're using. Another thing I've been playing with of late is using iptables' string matching functionality to block user agents of known attack vectors: 'sipcli', 'sipvicious', 'friendly-scanner', etc. This seems to work remarkably well, though what impact it has on net performance under load remains to be seen. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 secure fine tune - stop attack
Thank you all for your support, your suggestions are welcome. Thanks, On Thu, Sep 4, 2014 at 9:26 AM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 4/9/14 4:58 pm, Eric Wieling wrote: If we don't need to allow access from outside the USA we block access from all non-ARIN IP addresses by using iptables. This takes care of at least 80% of attacks. Likewise here (though RIPE rather than ARIN, since we're the other side of the pond). You can also take it a bit further: if, for example, you know what ISP(s) your dynamic clients are using, you can limit connections to the IP ranges those ISP(s) use - look up their ranges on he.net's BGP looking glass if you need to find out what ranges they're using. Another thing I've been playing with of late is using iptables' string matching functionality to block user agents of known attack vectors: 'sipcli', 'sipvicious', 'friendly-scanner', etc. This seems to work remarkably well, though what impact it has on net performance under load remains to be seen. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Special functionality for Secretary/Boss
We are currently migrating from a Nortel pbx to Asterisk and we have been able to convert most of the functions that people are used to but there is one I have no clear idea how to do. The scenario is: Boss calls secretary from outside the office to get connected to another outside destination. The secretary dials the destination and then trasfers call to the boss. When boss finishes with that person they want to send the call back to the secretary in order to make another connection or simply to talk to the secretary. The first part is not a problem, but after the boss finishes his call how can we send the call back to the secretary? I was thinking of using a conference room but how would the secretary know when the boss has finished? Anyone know how to handle this scenario? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] Special functionality for Secretary/Boss
Why can't you continue within the extension and dispatch whether the call failed or terminated? Simply make a second call. jg -- _ -- 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] Special functionality for Secretary/Boss
asterisk-users-boun...@lists.digium.com wrote on 09/04/2014 11:57:40 AM:
We are currently migrating from a Nortel pbx to Asterisk and we
have been able to convert most of the functions that people are used to
but there is one I have no clear idea how to do. The scenario is:
Boss calls secretary from outside the office to get connected to
another outside destination. The secretary dials the destination and
then trasfers call to the boss. When boss finishes with that person
they want to send the call back to the secretary in order to make
another connection or simply to talk to the secretary.
The first part is not a problem, but after the boss finishes his
call how can we send the call back to the secretary? I was thinking of
using a conference room but how would the secretary know when the boss
has finished? Anyone know how to handle this scenario?
I haven't tested this, but my initial thought would be to create a special
context or extension that the secretary could route through when doing the
call transfer. The Dial application could be called with the 'g' option to
continue the dialplan at the next priority when the call hangs up.
Something like a normal call transfer would just dial the number as
normal, but for the special transfer, you could prepend the dialed number
with a #.
For example (using a local US dialstring, change to fit your needs):
; This is a normal external call.
exten = _NXXNXXX,1,Dial(SIP/your_external_trunk/${EXTEN})
same = n,Hangup()
; This is a call that should be transfered back to the secretary's
extension when external call is finished
exten = _#NXXNXXX,1,NoOp(Special Dial for Boss/Secretary Transfer)
same = n,Dial(SIP/your_external_trunk/${EXTEN:1},,g)
; First call has ended, now we go back to the secretary)
same = n,Dial(SIP/1234)
same = n,Hangup()
That's at least where I would start with my testing and then develop the
solution from there.--
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Re: [asterisk-users] Special functionality for Secretary/Boss
Sounds like you are running FreePBX.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani
Sent: Thursday, September 04, 2014 6:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Special functionality for Secretary/Boss
Kevin,
With your dialplan with g option on external trunk, if the call finishes the
boss's leg of call also gets disconnected. So the next instruction would make a
call to secratary, however with no one on other end.
Mitul
On 04-Sep-2014 11:44 PM, Kevin Larsen
kevin.lar...@pioneerballoon.commailto:kevin.lar...@pioneerballoon.com wrote:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
wrote on 09/04/2014 11:57:40 AM:
We are currently migrating from a Nortel pbx to Asterisk and we
have been able to convert most of the functions that people are used to
but there is one I have no clear idea how to do. The scenario is:
Boss calls secretary from outside the office to get connected to
another outside destination. The secretary dials the destination and
then trasfers call to the boss. When boss finishes with that person
they want to send the call back to the secretary in order to make
another connection or simply to talk to the secretary.
The first part is not a problem, but after the boss finishes his
call how can we send the call back to the secretary? I was thinking of
using a conference room but how would the secretary know when the boss
has finished? Anyone know how to handle this scenario?
I haven't tested this, but my initial thought would be to create a special
context or extension that the secretary could route through when doing the call
transfer. The Dial application could be called with the 'g' option to continue
the dialplan at the next priority when the call hangs up. Something like a
normal call transfer would just dial the number as normal, but for the special
transfer, you could prepend the dialed number with a #.
For example (using a local US dialstring, change to fit your needs):
; This is a normal external call.
exten = _NXXNXXX,1,Dial(SIP/your_external_trunk/${EXTEN})
same = n,Hangup()
; This is a call that should be transfered back to the secretary's extension
when external call is finished
exten = _#NXXNXXX,1,NoOp(Special Dial for Boss/Secretary Transfer)
same = n,Dial(SIP/your_external_trunk/${EXTEN:1},,g)
; First call has ended, now we go back to the secretary)
same = n,Dial(SIP/1234)
same = n,Hangup()
That's at least where I would start with my testing and then develop the
solution from there.
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Re: [asterisk-users] Special functionality for Secretary/Boss
Kevin,
With your dialplan with g option on external trunk, if the call finishes
the boss's leg of call also gets disconnected. So the next instruction
would make a call to secratary, however with no one on other end.
Mitul
On 04-Sep-2014 11:44 PM, Kevin Larsen kevin.lar...@pioneerballoon.com
wrote:
asterisk-users-boun...@lists.digium.com wrote on 09/04/2014 11:57:40 AM:
We are currently migrating from a Nortel pbx to Asterisk and we
have been able to convert most of the functions that people are used to
but there is one I have no clear idea how to do. The scenario is:
Boss calls secretary from outside the office to get connected to
another outside destination. The secretary dials the destination and
then trasfers call to the boss. When boss finishes with that person
they want to send the call back to the secretary in order to make
another connection or simply to talk to the secretary.
The first part is not a problem, but after the boss finishes his
call how can we send the call back to the secretary? I was thinking of
using a conference room but how would the secretary know when the boss
has finished? Anyone know how to handle this scenario?
I haven't tested this, but my initial thought would be to create a special
context or extension that the secretary could route through when doing the
call transfer. The Dial application could be called with the 'g' option to
continue the dialplan at the next priority when the call hangs up.
Something like a normal call transfer would just dial the number as normal,
but for the special transfer, you could prepend the dialed number with a #.
For example (using a local US dialstring, change to fit your needs):
; This is a normal external call.
exten = _NXXNXXX,1,Dial(SIP/your_external_trunk/${EXTEN})
same = n,Hangup()
; This is a call that should be transfered back to the secretary's
extension when external call is finished
exten = _#NXXNXXX,1,NoOp(Special Dial for Boss/Secretary Transfer)
same = n,Dial(SIP/your_external_trunk/${EXTEN:1},,g)
; First call has ended, now we go back to the secretary)
same = n,Dial(SIP/1234)
same = n,Hangup()
That's at least where I would start with my testing and then develop the
solution from there.
--
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[asterisk-users] AstriCon Hackathon
Join a worldwide community of designers, developers, and communications technologists to to create, code, and design apps built on Asterisk and other communications APIs at the inaugural AstriCon Hackathon! Obligatory yet useful information: *When*: Wednesday, October 22nd at 8AM. A reception is on Tuesday, October 21st from 5 - 7 PM. *Where*: Red Rocks Casino in Las Vegas, Nevada (at AstriCon!) *How to apply*: Sign up at http://astriconhackathon.challengepost.com/ *Do I need a team?* If you have one, that's great! If not, you can post on the ChallengePost page (previously linked) or we can help you find one. *What can I win?* Prizes (no idea what, but I'm sure it will be suitably nifty) and the respect and admiration of your colleagues. If nothing else, you'll have a great time hacking on Asterisk and other communications APIs, while getting to lob whatever questions you may have at members of the Asterisk Development Team (myself included). More information about the hackathon can be found on the ChallengePost page or at http://www.asterisk.org/community/astricon-user-conference/hackathon See everyone in Las Vegas! 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
