Re: [asterisk-users] get start-time of all active calls
Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- 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] A few (simple?) questions
Hi all I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! My questions are regarding their setup - they have functionality split over several servers as follows (all running CentOS): Server1 MySQL Server2 Ruby on Rails + CSTele Server3 Asterisk 1.4.19 + STUN #1 Server4 Trunk (Asterisk 1.4.19) + STUN #2 Server5 Apache ActiveMQ The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. My questions are as follows: - STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? - Why have a separate Asterisk server for the trunk? - Is the Apache Message Queue server necessary? - My info says that server 2 is running CSTele but I have been unable to find a process or program that matches this (except for a comment in a daemon, ast_ami_events.rb, running on Rails server). Can anybody tell me what CSTele might be? Many thanks Binni ITAnet Kirkestien 20 9230 Svenstrup Telefon: 3020 0868 Email: bi...@itanet.numailto:i...@itanet.nu WWW: http://www.itanet.nuhttp://www.itanet.nu/ [cid:image001.gif@01CCBA43.F35D3990] inline: image001.gif-- _ -- 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] get start-time of all active calls
Hi, Not sure why you didnt get it, when I did thta command for originator channel it showed me the CDR variables list which included CDR Variables: level 1: dnid= level 1: clid=XXX level 1: src= level 1: dst= level 1: dcontext=SIP-incoming level 1: channel= level 1: dstchannel= level 1: lastapp=Dial level 1: lastdata=SIP/ *level 1: start=2011-12-14 09:15:54* level 1: answer=2011-12-14 09:16:01 level 1: duration=11 level 1: billsec=4 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1323854154.856 level 1: linkedid=1323854154.856 level 1: sequence=1096 Thats valid for an ongoing bridged call-initiator side only. Regards, Sammy On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh -- Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- 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] Realtime Registration
On Tue, 2011-12-13 at 16:32 -0800, Edwin Lam wrote: On 12/10/11 9:54 PM, Takehiro Matsushima wrote: I'd configured realtime registration, but configuration was not applied when I changed a row of sippeers table. To apply, 'sip reload' was needed (in Asterisk 1.8.0). or you can 'sip prune realtime extension' This is fundamental to understanding realtime registrations. When a peer tries to register the realtime cache is populated from the contents of the sip table, for that row, as it is at that point in time. The peer will then always use the configuration as it is in the cache. To make changes to the configuration of a peer not only do you have to make a change to the table entry, you then need to flush the realtime cache for that peer which is done by the command given above. The peer will then try to reconnect and the realtime cache will be populated from the updated DB entry. Hope this helps Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] get start-time of all active calls
In article CAJUJwthT=mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com, Sammy Govind govoi...@gmail.com wrote: Hi, Not sure why you didnt get it, when I did thta command for originator channel it showed me the CDR variables list which included That's from show channel, not sip show channel. Cheers Tony CDR Variables: level 1: dnid= level 1: clid=XXX level 1: src= level 1: dst= level 1: dcontext=SIP-incoming level 1: channel= level 1: dstchannel= level 1: lastapp=Dial level 1: lastdata=SIP/ *level 1: start=2011-12-14 09:15:54* level 1: answer=2011-12-14 09:16:01 level 1: duration=11 level 1: billsec=4 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1323854154.856 level 1: linkedid=1323854154.856 level 1: sequence=1096 Thats valid for an ongoing bridged call-initiator side only. Regards, Sammy On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh -- Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- 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 -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- -- _ -- 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 -=-=-=-=-=- -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
Re: [asterisk-users] get start-time of all active calls
oops, you got it. On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield t...@softins.co.ukwrote: In article CAJUJwthT= mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com, Sammy Govind govoi...@gmail.com wrote: Hi, Not sure why you didnt get it, when I did thta command for originator channel it showed me the CDR variables list which included That's from show channel, not sip show channel. Cheers Tony CDR Variables: level 1: dnid= level 1: clid=XXX level 1: src= level 1: dst= level 1: dcontext=SIP-incoming level 1: channel= level 1: dstchannel= level 1: lastapp=Dial level 1: lastdata=SIP/ *level 1: start=2011-12-14 09:15:54* level 1: answer=2011-12-14 09:16:01 level 1: duration=11 level 1: billsec=4 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1323854154.856 level 1: linkedid=1323854154.856 level 1: sequence=1096 Thats valid for an ongoing bridged call-initiator side only. Regards, Sammy On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh -- Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- 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 -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- -- _ -- 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 -=-=-=-=-=- -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org --
Re: [asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)
On Thu, 1 Dec 2011 14:09:29 +0300, James Mutuku listmut...@gmail.com wrote: I have worked with bare asterisk + freepbx before. the mypbx was just an example but my reference to appliances as a whole. The appliances seem to have lower entry costs. Appliances have less RAM + storage, so you'll have to make sure they're OK for what you're trying to do. Also, they usually use non-x86 chips, which means you're restricted to the OS + add-ons available for that platform. www.voip-info.org/wiki/view/Asterisk+Appliances www.astlinux.org www.smallnetbuilder.com/multimedia-voip/multimedia-voip-features/31208-how-to-build-asterisk-appliances-on-the-cheap -- _ -- 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] get start-time of all active calls
finally I got it with 'core show channel' channel-id thanks for your support. Date: Wed, 14 Dec 2011 15:11:49 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls oops, you got it. On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield t...@softins.co.uk wrote: In article CAJUJwthT=mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com, Sammy Govind govoi...@gmail.com wrote: Hi, Not sure why you didnt get it, when I did thta command for originator channel it showed me the CDR variables list which included That's from show channel, not sip show channel. Cheers Tony CDR Variables: level 1: dnid= level 1: clid=XXX level 1: src= level 1: dst= level 1: dcontext=SIP-incoming level 1: channel= level 1: dstchannel= level 1: lastapp=Dial level 1: lastdata=SIP/ *level 1: start=2011-12-14 09:15:54* level 1: answer=2011-12-14 09:16:01 level 1: duration=11 level 1: billsec=4 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1323854154.856 level 1: linkedid=1323854154.856 level 1: sequence=1096 Thats valid for an ongoing bridged call-initiator side only. Regards, Sammy On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh -- Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- 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 -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- -- _ -- 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 -=-=-=-=-=- -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play:
Re: [asterisk-users] A few (simple?) questions
On 14-12-11 10:18, Brynjolfur Thorvardsson wrote: Hi all I’ve been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! My questions are regarding their setup – they have functionality split over several servers as follows (all running CentOS): Server1 MySQL Server2 Ruby on Rails + CSTele Server3 Asterisk 1.4.19 + STUN #1 Server4 Trunk (Asterisk 1.4.19) + STUN #2 Server5 Apache ActiveMQ The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. My questions are as follows: -STUN server – is it necessary (given that there are many free STUN servers on the Internet), and why two? Why would you want to rely on a free stun server which can disappear anytime when offering commercial services? I would also deploy my own stun servers for paying customers. -Why have a separate Asterisk server for the trunk? No idea. Maybe the question could be: why have two Asterisk servers? Perhaps for for redundancy/failover? -Is the Apache Message Queue server necessary? No idea. I know BigBlueButton uses Apache MQ Asterisk but I don't know the specifics. -My info says that server 2 is running CSTele but I have been unable to find a process or program that matches this (except for a comment in a daemon, ast_ami_events.rb, running on Rails server). Can anybody tell me what CSTele might be? No idea. Good luck! Regards, 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
[asterisk-users] Asterisk 1.4.x segfaulting daily
Hello list,
An Asterisk installation that was doing fine suddenly stared segfaulting
a couple of times per day. I enabled all the logging and debugging to
try to find a pattern but there was too much information to see exactly
where it broke. So I enabled core dump and did backtraces and all of
them seem to break on ast_setstate, setting the state to AST_STATE_DOWN.
That's pretty much the only thing I can make of it, don't even know if
that's correct.
Does anyone have any ideas on why this is happening? The backtrace is
attached.
P.S.: I've switched the whole hardware already, including the BRI card
(B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and
mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1.
Best regards,
Paulo Santos
Core was generated by `/usr/sbin/asterisk'.
Program terminated with signal 11, Segmentation fault.
[New process 21726]
[New process 24376]
[New process 24375]
[New process 24374]
[New process 24371]
[New process 24344]
[New process 23560]
[New process 22868]
[New process 22329]
[New process 22327]
[New process 22325]
[New process 22324]
[New process 22323]
[New process 22322]
[New process 22321]
[New process 22320]
[New process 22319]
[New process 22318]
[New process 22317]
[New process 22316]
[New process 22315]
[New process 22259]
[New process 22208]
[New process 22203]
[New process 22185]
[New process 22184]
[New process 22160]
[New process 21515]
[New process 21725]
[New process 21687]
[New process 21686]
[New process 21685]
[New process 21681]
[New process 21659]
[New process 21658]
[New process 21648]
[New process 21647]
[New process 21609]
[New process 21594]
[New process 21542]
[New process 21540]
[New process 21516]
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at
/usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
37 return (!s || (*s == '\0'));
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at
/usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
name =
mISDN/4\000u\000ݴ��\177�\020\000@�\b(@�H0ݴf\211q�\020\000@�\b(@�\000\000@�Хe�\b(@�X\b@�h0ݴ\203\225a�P[
3] \000\000\000\000\000
#1 0xb561975d in release_chan (ch=0xb3400858, bc=0x88e8f5c) at
chan_misdn.c:3750
ast = (struct ast_channel *) 0xb3401c00
#2 0xb5622275 in cb_events (event=EVENT_CLEANUP, bc=0x88e8f5c, user_data=0x0)
at chan_misdn.c:4845
msn_valid = -1287644160
held_ch = value optimized out
ch = (struct chan_list *) 0xb3400858
__PRETTY_FUNCTION__ = cb_events
#3 0xb5632d9f in handle_cr (stack=0x88e82d8, frm=value optimized out) at
misdn/isdn_lib.c:1684
channel = 255
bc = (struct misdn_bchannel *) 0x88e8f5c
dummybc = {send_lock = 0xb67feff4, dummy = -1260570753, nt =
-1260572104, pri = -1234083825, port = -1260572068, b_stid = -1260571776,
layer_id = -1260570753, layer = -1234274741, need_disconnect = -1233129484,
need_release = -1260572068, need_release_complete = -1260571776,
dec = -1260571832, l3_id = -1234111388, pid = -1260572068, ces = -1251638304,
restart_channel = -1260570700, channel = -1260571776,
channel_preselected = 0, in_use = -1260571908, last_used = {tv_sec = 1023,
tv_usec = -72515583}, cw = -1260571776, addr = -1260571776,
bframe = 0xb4dd3380 handle_frm: frm-addr:42000303 frm-prim:3f182\n,
bframe_len = -1260571776, time_usec = -1260571729,
astbuf = 0xb4dd377f, misdnbuf = 0xb4dd3380, te_choose_channel = -1260570753,
early_bconnect = 0, dtmf = 0, send_dtmf = 0,
need_more_infos = 0, sending_complete = 0, nodsp = 1635021600, nojitter = 0,
dnumplan = NUMPLAN_UNKNOWN, rnumplan = 1308622848,
onumplan = NUMPLAN_UNKNOWN, cpnnumplan = NUMPLAN_UNINITIALIZED,
progress_coding = 824193585, progress_location = 942881334,
progress_indicator = 3617594, fac_in = {Function = Fac_GetSupportedServices,
u = {Listen = {NotificationMask = 21}, Suspend = {
CallIdentity = \025\000\000\000\000\000\000\000\000\000\000},
Resume = {
CallIdentity = \025\000\000\000\000\000\000\000\000\000\000},
CFActivate = {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000,
ForwardedToNumber = @�\177�\000\000\000\000�wa�\0203ݴ,
ForwardedToSubaddress = \000\004\000\000�ze�7ݴ@�\177�}, CFDeactivate
= {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000},
CFInterrogateParameters = {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000},
CFInterrogateNumbers = {Handle = 21}, CDeflection = {
PresentationAllowed = 21, DeflectedToNumber =
\000\000\000\000\000\000\000\000\000\000Х,
DeflectedToSubaddress = e�\001\000\000\000@�\177�\000\000\000\000�w},
AOCDchu = {chargeNotAvailable = 21, freeOfCharge = 0,
recordedUnits = 0, typeOfChargingInfo = -1, billingId = 0}, AOCDcur =
{chargeNotAvailable
Re: [asterisk-users] Asterisk 1.4.x segfaulting daily
On 14 December 2011 12:56, Paulo Santos paulo.r.san...@sapo.pt wrote: Hello list, An Asterisk installation that was doing fine suddenly stared segfaulting a couple of times per day. I enabled all the logging and debugging to try to find a pattern but there was too much information to see exactly where it broke. So I enabled core dump and did backtraces and all of them seem to break on ast_setstate, setting the state to AST_STATE_DOWN. That's pretty much the only thing I can make of it, don't even know if that's correct. Does anyone have any ideas on why this is happening? The backtrace is attached. P.S.: I've switched the whole hardware already, including the BRI card (B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1. If I was guessing, I'd say that the channel structure that is being modified by the ast_setstate() call is incomplete, and contains some garbage pointers. If I was guessing further, I'd say that the callerID pointers are the most likely candidate - Does the issue happen when a caller-id withheld call is hung-up? hung-up before being answered perhaps? You'd need to add some debug reporting into ast_setstate() to know for sure. Just my 2p - 1.4.42 is an old version, so the chance of a solid answer is fairly low. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.x segfaulting daily
On 14-12-11 13:56, Paulo Santos wrote: Hello list, An Asterisk installation that was doing fine suddenly stared segfaulting a couple of times per day. I enabled all the logging and debugging to try to find a pattern but there was too much information to see exactly where it broke. So I enabled core dump and did backtraces and all of them seem to break on ast_setstate, setting the state to AST_STATE_DOWN. That's pretty much the only thing I can make of it, don't even know if that's correct. Does anyone have any ideas on why this is happening? The backtrace is attached. P.S.: I've switched the whole hardware already, including the BRI card (B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1. If the suggestion from Steve Davies doesn't work out for you then my suggestion would be to try out the latest DAHDI libpri with the latest Asterisk 1.8 because those versions have built-in support for the 4x BRI HFC chipset which can be found on the Digium b410p and the Openvox B400P. This way you no longer need mISDN V1 and have recent versions with tons of bugs fixed. Here are instructions from Openvox: http://wiki.openvox.cn/index.php/OpenVox_B400P_User_Manual_for_dahdi Please note that in the instructions they use older versions. I would use the latest DAHDI, libpri (don't forget this one) and asterisk 1.8 available here: https://www.asterisk.org/downloads Regards, 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] A few (simple?) questions
Hi, thanks for your answer. I suppose that both the STUN servers and ActiveMQ are there to give a better/more reliable service which is obviously a good idea. From trying to find out some more on the Internet I get the idea that CSTele might have something to do with Circuit Switching. I am guessing that the CSTele server establishes a virtual switching circuit to the queue server and trunk server, possibly through a separate network card (servers 3,4 and 5 all have an extra ethernet card without fixed IP address). Regards Binni -Oprindelig meddelelse- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Patrick Lists Sendt: 14. december 2011 13:45 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions On 14-12-11 10:18, Brynjolfur Thorvardsson wrote: Hi all I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! My questions are regarding their setup - they have functionality split over several servers as follows (all running CentOS): Server1 MySQL Server2 Ruby on Rails + CSTele Server3 Asterisk 1.4.19 + STUN #1 Server4 Trunk (Asterisk 1.4.19) + STUN #2 Server5 Apache ActiveMQ The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. My questions are as follows: -STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? Why would you want to rely on a free stun server which can disappear anytime when offering commercial services? I would also deploy my own stun servers for paying customers. -Why have a separate Asterisk server for the trunk? No idea. Maybe the question could be: why have two Asterisk servers? Perhaps for for redundancy/failover? -Is the Apache Message Queue server necessary? No idea. I know BigBlueButton uses Apache MQ Asterisk but I don't know the specifics. -My info says that server 2 is running CSTele but I have been unable to find a process or program that matches this (except for a comment in a daemon, ast_ami_events.rb, running on Rails server). Can anybody tell me what CSTele might be? No idea. Good luck! Regards, 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 -- _ -- 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 1.4.x segfaulting daily
Hello, Thank you all for the replies. Steve Davies wrote: If I was guessing, I'd say that the channel structure that is being modified by the ast_setstate() call is incomplete, and contains some garbage pointers. If I was guessing further, I'd say that the callerID pointers are the most likely candidate - Does the issue happen when a caller-id withheld call is hung-up? hung-up before being answered perhaps? It was an outgoing call that tried to call through the port 2, then 1 and finally 3. The third port has a quite different debug output than the other 2. Maybe it's a problem on that connection, appears to be common on all segfaults. Apparently that third port is something of a strange group of BRI lines between that one and the line on the second port, but behaves differently. I'll try to find out more about it. Patrick Lists wrote: If the suggestion from Steve Davies doesn't work out for you then my suggestion would be to try out the latest DAHDI libpri with the latest Asterisk 1.8 because those versions have built-in support for the 4x BRI HFC chipset which can be found on the Digium b410p and the Openvox B400P. This way you no longer need mISDN V1 and have recent versions with tons of bugs fixed. Unfortunately I can't do that, at least not now. I will, however, migrate it eventually to either mISDN v2 or Dahdi, depending on the state of Dahdi then. P.S.: Attached the log. Best regards, Paulo Santos [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 333232837-5062-310@192.168.0.8 Their Tag 1036797295 Our tag: as5b7769e2 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1693981358-5068-505@192.168.0.7 Their Tag 692402733 Our tag: as170cc25e [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1394539361-5064-828@192.168.0.7 Their Tag 1627163612 Our tag: as5f15bf50 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1708030692-5060-122@192.168.0.8 Their Tag 52015999 Our tag: as24b80c2d [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to Off [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to Off [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Allocating new SIP dialog for 1547819775-5062-295@192.168.0.7 - INVITE (With RTP) [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 0.0.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 192.168.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 10.0.0.0 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = Found Their Call ID: 1547819775-5062-295@192.168.0.7 Their Tag 2074339809 Our tag: as2515e4b3 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received ACK (6) - Command in SIP ACK [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Stopping retransmission on '1547819775-5062-295@192.168.0.7' of Response 2940: Match Found [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = Found Their Call ID: 1547819775-5062-295@192.168.0.7 Their Tag 2074339809 Our tag: as2515e4b3 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 0.0.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 192.168.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 10.0.0.0 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP o=11 8002 8000 IN IP4 192.168.0.7... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.7... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK. [Dec 12 16:38:36] DEBUG[22160]
Re: [asterisk-users] A few (simple?) questions
On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: ** I’ve been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. Sounds way over-built, but since we don't know the intent of the architecture nor all the features expected, hard to say. - **STUN server – is it necessary (given that there are many free STUN servers on the Internet), and why two? I don't believe so. **- **Why have a separate Asterisk server for the trunk? Can't think of any reason. **- **Is the Apache Message Queue server necessary? Necessary is not something that can be answered. In their environment as programmed, probably. In general, can an Asterisk server run without it? Yes. A low-end single x86 server can easily support hundreds of endpoints and dozens of concurrent calls, with all Asterisk services running on a single server. ** Do you have Asterisk expertise already? RoR, SQL, other telephony...? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] VoiceMail and IMAP
Any thoughts on what could be causing this ? -- Thanks, Phil - Original Message - Okay, though removing the space and reloading the module still throws the same error messages. -- Thanks, Phil - Original Message - Generally speaking, no. if you need the space, use quotes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: Monday, December 12, 2011 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoiceMail and IMAP Hmmm, just tried leaving a voicemail on a new mailbox where the imapfolder contains a space in the name and it errors; so that could be the cause of it all. Is is valid to have a space in an IMAP folder name ? -- Thanks, Phil - Original Message - 1.8.7.0 ... am using Zimbra as the backend IMAP storage. -- Thanks, Phil - Original Message - On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: From what version? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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 -- _ -- 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] A few (simple?) questions
Hi Carlos and thanks for your answer. To begin with: I am a noob in all telephony/asterisk/ror fields, coming from a Classic ASP/MS background! I've been nosing around in RoR and Asterisk for the last month or so and have managed to create several RoR sites and to get an Asterisk server up and running so me and my boss can phone each other using softphone on a smartphone. So, yes it's going to be fun! And again, thanks for your answer. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:13 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu wrote: I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. Sounds way over-built, but since we don't know the intent of the architecture nor all the features expected, hard to say. - STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? I don't believe so. - Why have a separate Asterisk server for the trunk? Can't think of any reason. - Is the Apache Message Queue server necessary? Necessary is not something that can be answered. In their environment as programmed, probably. In general, can an Asterisk server run without it? Yes. A low-end single x86 server can easily support hundreds of endpoints and dozens of concurrent calls, with all Asterisk services running on a single server. Do you have Asterisk expertise already? RoR, SQL, other telephony...? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] A few (simple?) questions
Getting involved in an existing, and possibly broken system is the wrong way to start with Asterisk. I know, because that's how my career in VoIP started. I had to unlearn a lot of poor practices I learned from that system. But anyway without prior documentation or the ability to get the original design intention, I think your next step is to go right back to the beginning, and gather the user requirements and create a design. Then see if it was solved properly, or you need to start over, or what. Without the basics I don't think you can answer the questions you had. Once you know what was needed and why it was custom-written, you'll probably have all those answers. Just know that in its basic form, to process calls for a normal company, nothing is needed other than one Asterisk server. Everything else is extra, which may or may not be warranted. I've seen a number of deployments that seemed geared more towards making a very profitable complex custom system than just giving the customer the best value. Asterisk is a particularly noob-unfriendly product with a lot of pitfalls and relatively poor documentation. Don't go into it lightly, and always be aware that doing it wrong results in anything from system failures to thousands of dollars in toll fraud costs. On Wed, Dec 14, 2011 at 8:38 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: Hi Carlos and thanks for your answer. To begin with: I am a noob in all telephony/asterisk/ror fields, coming from a Classic ASP/MS background! I’ve been nosing around in RoR and Asterisk for the last month or so and have managed to create several RoR sites and to get an Asterisk server up and running so me and my boss can phone each other using softphone on a smartphone. ** ** So, yes it’s going to be fun! And again, thanks for your answer. ** ** ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *Carlos Alvarez *Sendt:* 14. december 2011 16:13 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* Re: [asterisk-users] A few (simple?) questions ** ** On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.nu wrote: I’ve been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! ** ** Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. ** ** The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. ** ** Sounds way over-built, but since we don't know the intent of the architecture nor all the features expected, hard to say. - STUN server – is it necessary (given that there are many free STUN servers on the Internet), and why two? ** ** I don't believe so. - Why have a separate Asterisk server for the trunk? Can't think of any reason. - Is the Apache Message Queue server necessary? Necessary is not something that can be answered. In their environment as programmed, probably. In general, can an Asterisk server run without it? Yes. A low-end single x86 server can easily support hundreds of endpoints and dozens of concurrent calls, with all Asterisk services running on a single server. Do you have Asterisk expertise already? RoR, SQL, other telephony...? ** ** -- Carlos Alvarez TelEvolve 602-889-3003 ** ** ** ** -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] A few (simple?) questions
Hi Carlos and thanks for the advice. I agree with you wholeheartedly but I'm not sure if I have much choice in the matter. The system was originally designed to offer PBX services to private clinics and currently handles between 10 and 20, with 70 phone numbers. The guys I work for want to expand into other market segments here in Denmark and my job is to re-install the system on some new servers and start making changes. The code is not very well written, the original developers have totally misunderstood the RVM model in Rails and the Asterix config files are full of unused code and example code. There is also some very sloppy version control in the Rails/Adhearsion files and absolutely no regression testing. But, hey, it seems to work! I would like to start from fresh and re-develop the system, I am not at all confident of being able to just lift the code from the current servers and copy/paste it all onto some new ones and expect it to work. Your solid advice might help me make the case for a fresh start, but whichever way it goes, at least I'll be kept busy ... Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:58 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions Getting involved in an existing, and possibly broken system is the wrong way to start with Asterisk. I know, because that's how my career in VoIP started. I had to unlearn a lot of poor practices I learned from that system. But anyway without prior documentation or the ability to get the original design intention, I think your next step is to go right back to the beginning, and gather the user requirements and create a design. Then see if it was solved properly, or you need to start over, or what. Without the basics I don't think you can answer the questions you had. Once you know what was needed and why it was custom-written, you'll probably have all those answers. Just know that in its basic form, to process calls for a normal company, nothing is needed other than one Asterisk server. Everything else is extra, which may or may not be warranted. I've seen a number of deployments that seemed geared more towards making a very profitable complex custom system than just giving the customer the best value. Asterisk is a particularly noob-unfriendly product with a lot of pitfalls and relatively poor documentation. Don't go into it lightly, and always be aware that doing it wrong results in anything from system failures to thousands of dollars in toll fraud costs. On Wed, Dec 14, 2011 at 8:38 AM, Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu wrote: Hi Carlos and thanks for your answer. To begin with: I am a noob in all telephony/asterisk/ror fields, coming from a Classic ASP/MS background! I've been nosing around in RoR and Asterisk for the last month or so and have managed to create several RoR sites and to get an Asterisk server up and running so me and my boss can phone each other using softphone on a smartphone. So, yes it's going to be fun! And again, thanks for your answer. Fra: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:13 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu wrote: I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. Sounds way over-built, but since we don't know the intent of the architecture nor all the features expected, hard to say. - STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? I don't believe so. - Why have a separate Asterisk server for the trunk? Can't think of any reason. - Is the Apache Message Queue server necessary? Necessary is not something that can be answered. In their environment as programmed, probably. In general, can an Asterisk server run without it? Yes. A low-end single x86 server can easily support hundreds of endpoints and dozens of concurrent calls, with all Asterisk services running on a single server. Do you have Asterisk expertise already? RoR, SQL, other
Re: [asterisk-users] A few (simple?) questions
Please feel free to pass this along: DON'T DO IT! Taking questionable code, from what appears to be a questionable relationship, and then trying to extend its life is probably the craziest way to go about this. You, personally, are in for a steep learning curve on this. Having worked with Asterisk for six years now, I can look back and see that jumping into complex projects with it at the beginning would have led to many problems. On Wed, Dec 14, 2011 at 9:22 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: Hi Carlos and thanks for the advice. I agree with you wholeheartedly but I’m not sure if I have much choice in the matter. The system was originally designed to offer PBX services to private clinics and currently handles between 10 and 20, with 70 phone numbers. The guys I work for want to expand into other market segments here in Denmark and my job is to re-install the system on some new servers and start making changes. ** ** The code is not very well written, the original developers have totally misunderstood the RVM model in Rails and the Asterix config files are full of unused code and example code. There is also some very sloppy version control in the Rails/Adhearsion files and absolutely no regression testing. But, hey, it seems to work! ** ** I would like to start from fresh and re-develop the system, I am not at all confident of being able to just lift the code from the current servers and copy/paste it all onto some new ones and expect it to work. Your solid advice might help me make the case for a fresh start, but whichever way it goes, at least I’ll be kept busy ... ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *Carlos Alvarez *Sendt:* 14. december 2011 16:58 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* Re: [asterisk-users] A few (simple?) questions ** ** Getting involved in an existing, and possibly broken system is the wrong way to start with Asterisk. I know, because that's how my career in VoIP started. I had to unlearn a lot of poor practices I learned from that system. ** ** But anyway without prior documentation or the ability to get the original design intention, I think your next step is to go right back to the beginning, and gather the user requirements and create a design. Then see if it was solved properly, or you need to start over, or what. Without the basics I don't think you can answer the questions you had. Once you know what was needed and why it was custom-written, you'll probably have all those answers. Just know that in its basic form, to process calls for a normal company, nothing is needed other than one Asterisk server. Everything else is extra, which may or may not be warranted. I've seen a number of deployments that seemed geared more towards making a very profitable complex custom system than just giving the customer the best value. ** ** Asterisk is a particularly noob-unfriendly product with a lot of pitfalls and relatively poor documentation. Don't go into it lightly, and always be aware that doing it wrong results in anything from system failures to thousands of dollars in toll fraud costs. ** ** On Wed, Dec 14, 2011 at 8:38 AM, Brynjolfur Thorvardsson bi...@itanet.nu wrote: Hi Carlos and thanks for your answer. To begin with: I am a noob in all telephony/asterisk/ror fields, coming from a Classic ASP/MS background! I’ve been nosing around in RoR and Asterisk for the last month or so and have managed to create several RoR sites and to get an Asterisk server up and running so me and my boss can phone each other using softphone on a smartphone. So, yes it’s going to be fun! And again, thanks for your answer. *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *Carlos Alvarez *Sendt:* 14. december 2011 16:13 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* Re: [asterisk-users] A few (simple?) questions On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.nu wrote: I’ve been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. Sounds way over-built, but since we don't know the intent of the architecture nor all the features expected, hard to say. - STUN server – is
Re: [asterisk-users] A few (simple?) questions
You are 110% correct Carlos, but Im sure B.T. likes to eat. We all have to do things we dont like. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Wednesday, December 14, 2011 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] A few (simple?) questions Please feel free to pass this along: DON'T DO IT! Taking questionable code, from what appears to be a questionable relationship, and then trying to extend its life is probably the craziest way to go about this. You, personally, are in for a steep learning curve on this. Having worked with Asterisk for six years now, I can look back and see that jumping into complex projects with it at the beginning would have led to many problems. On Wed, Dec 14, 2011 at 9:22 AM, Brynjolfur Thorvardsson bi...@itanet.nu wrote: Hi Carlos and thanks for the advice. I agree with you wholeheartedly but Im not sure if I have much choice in the matter. The system was originally designed to offer PBX services to private clinics and currently handles between 10 and 20, with 70 phone numbers. The guys I work for want to expand into other market segments here in Denmark and my job is to re-install the system on some new servers and start making changes. The code is not very well written, the original developers have totally misunderstood the RVM model in Rails and the Asterix config files are full of unused code and example code. There is also some very sloppy version control in the Rails/Adhearsion files and absolutely no regression testing. But, hey, it seems to work! I would like to start from fresh and re-develop the system, I am not at all confident of being able to just lift the code from the current servers and copy/paste it all onto some new ones and expect it to work. Your solid advice might help me make the case for a fresh start, but whichever way it goes, at least Ill be kept busy ... Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:58 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions Getting involved in an existing, and possibly broken system is the wrong way to start with Asterisk. I know, because that's how my career in VoIP started. I had to unlearn a lot of poor practices I learned from that system. But anyway without prior documentation or the ability to get the original design intention, I think your next step is to go right back to the beginning, and gather the user requirements and create a design. Then see if it was solved properly, or you need to start over, or what. Without the basics I don't think you can answer the questions you had. Once you know what was needed and why it was custom-written, you'll probably have all those answers. Just know that in its basic form, to process calls for a normal company, nothing is needed other than one Asterisk server. Everything else is extra, which may or may not be warranted. I've seen a number of deployments that seemed geared more towards making a very profitable complex custom system than just giving the customer the best value. Asterisk is a particularly noob-unfriendly product with a lot of pitfalls and relatively poor documentation. Don't go into it lightly, and always be aware that doing it wrong results in anything from system failures to thousands of dollars in toll fraud costs. On Wed, Dec 14, 2011 at 8:38 AM, Brynjolfur Thorvardsson bi...@itanet.nu wrote: Hi Carlos and thanks for your answer. To begin with: I am a noob in all telephony/asterisk/ror fields, coming from a Classic ASP/MS background! Ive been nosing around in RoR and Asterisk for the last month or so and have managed to create several RoR sites and to get an Asterisk server up and running so me and my boss can phone each other using softphone on a smartphone. So, yes its going to be fun! And again, thanks for your answer. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:13 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.nu wrote: Ive been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms
