Re: [asterisk-users] ADSL Load Balancing
On Tue, 2 Nov 2010, Dan Journo wrote: Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Get a Draytek 2820 router/modem and a Vigor 120 ADSL modem and do it in the 2820. Much easier. (The 2820 has one ADSL port and one WAN port which will run pppoe to the 120 modem) However if you're up for it, then 2 separate ADSL modem/routers and read this: http://lartc.org/howto/ specifically section 4. Gordon -- _ -- 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] inbound call issue...
Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: --- SIP read from UDP:147.135.32.221:5060 --- INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Contact: sip:4144038...@147.135.32.221:5060 Supported: 100rel Max-Forwards: 69 Content-Length: 308 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.248 t=0 0 m=audio 15502 RTP/AVP 0 18 8 96 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 - [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- (11 headers 14 lines) --- [Nov 3 02:08:40] VERBOSE[7207] netsock.c: == Using SIP RTP CoS mark 5 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Sending to 147.135.32.221 : 5060 (NAT) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Using INVITE request as basis request - 31007e...@147.135.32.221 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Found peer 'trunk_1' for '4144038968' from 147.135.32.221:5060 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- Reliably Transmitting (NAT) to 147.135.32.221:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281-;received=147.135.32.221 From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22;tag=as4fffe111 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2dd58be8 Content-Length: 0 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Scheduling destruction of SIP dialog '31007e...@147.135.32.221' in 32000 ms (Method: INVITE) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- SIP read from UDP:147.135.32.221:5060 --- ACK sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 ACK From: Wi Msip:number f...@147.135.32.221;user=phone;tag=9bbc To: usernamesip:s...@216.26.109.22;tag=as4fffe111 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length:0 Here's the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register = 6087294351:sip password@sip.broadvoice.com [trunk_1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=6087294351 secret=sip password username=6087294351 insecure=very context=DID_trunk_1 authname=6087294351 dtmfmode=inband dtmf=inband canreinvite=no [guest] type=friend host=dynamic canreinvite=no context=DID_trunk_1 -- _ -- 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 linphone call dropping by itself
hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two linphones register with a name which is defined as dynamic in sip.conf the call terminates on the caller's side, while the callee is still connected, and I have to force the termination on that side. I'm using asterisk 1.8.0 and linphone 3.99 I really don't know how to investigate further... a capture on sip ports just shows that on the 25th ack packet the other side answers with a BYE instead of with an OK SDP packet. TIA, Matteo -- _ -- 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] ADSL Load Balancing
On 11/03/2010 03:49 AM, Gordon Henderson wrote: I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? I use the SG560 (http://www.snapgear.com/index.cfm?skey=1557) to do this. It handles two WAN connections (going to your ADSL modems). I set the routing policies so that VOIP goes on one link by default, and everything else on the other. If one link goes down, everything will be routed on the remaining link. (Unfortunately, it doesn't seem to revert to the default state after the downed link recovers, so I have to add some reboot-modems-after-recovery scripts in a cron job to make things recovery in an ideal way.) I think you can do the same with the Cisco RV016, which is cheaper, but the documentation is poor. - Mike -- _ -- 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, VoIP and Samsung iDCS100
Thanks everyone for your replies so far. I've pretty much concluded that going for a full Asterisk solution is the best longer term solution and that's what I'll do. We're moving office before May so that's the perfect time to put in a new phone system. But, I need to implement something quick-time just to buy me some time to do a full Asterisk solution. With that in mind, a couple of questions below: Gordon Henderson said at 02/11/2010 16:39: On Tue, 2 Nov 2010, Ronny Adsetts wrote: 1. Add analogue card(s) to the computer to run Asertisk and treat them as analogue extensions in the Samsung. Statically route each extension to a VoIP handset/user. So incoming via ISDN, Samsung converts to analogue, PC converts to VoIP and then out again - it'll work (maybe), but it's a huge waste of resources. It is a waste of resources I agree but might be the easiest way forward. As I mentioned above, I need a quick and dirty short term solution to buy me some time to scrap the Samsung and replace with Asterisk. I have an analogue extensions card in the Samsung that we currently use for a fax and answer-phone for one of our numbers (long story). What hardware would I need in the Asterisk so I could hook up some analogue extensions? Am I right in thinking I need something like an FXO/FXS card? I imagine I could then route (effectively hard-wire) those extensions direct each to a specific SIP phone using Asterisk? Thanks again for all your help so far. Ronny -- Ronny Adsetts Technical Director Amazing Internet Ltd, London t: +44 20 8607 9535 f: +44 20 8607 9536 w: www.amazinginternet.com Registered office: UK House, 82 Heath Road, Twickenham TW1 4BW Registered in England. Company No. 4042957 signature.asc Description: OpenPGP digital signature -- _ -- 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 and SIP a Provider in Brazil
I have sent an e-mail to this list (awaiting moderator approval by the size) talking about some difficult to make calls with a SIP Provider in Brazil. I'm new at this list and have no sure if I have posted my question in the right place. If this is not the channel to make this kind of question about this issue, I'm sorry but want to ask if anyone can indicate the correct place . If you are brazilian, enter in brasilian comunity of Asterisk [1]. Ther you will have the best information about Voip providers in Brazil. The people of AsteriskBrasil have a lots of experience with the providers in Brazil. [1] http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil Att, -- Rodrigo Lang, Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- 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, VoIP and Samsung iDCS100
On Wed, Nov 03, 2010 at 12:05:51PM +, Ronny Adsetts wrote: What hardware would I need in the Asterisk so I could hook up some analogue extensions? Am I right in thinking I need something like an FXO/FXS card? Yes, this ought to work. If you're plugging phones into the Samsung it's providing an FXS interface, so you'll need an FXO interface to talk to that; if you want to connect those analogue phones to Asterisk, you'll also need FXS interfaces (though as a short-term fix it would probably be easier to leave them plumbed directly into the Samsung box). Getting four modules (each of which can be FXO or FXS) on a single card is pretty easy (I use an OpenVox A400P from voipon, following recommendations on this list). You could then connect (some combination of) analogue channels to (some combination of) SIP phones, and vice versa to allow outward dialling. Once you build the Asterisk-only system, you can use the FXO modules to connect to analogue PSTN lines (assuming you have a use for this). Roger signature.asc Description: Digital signature -- _ -- 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] inbound call issue...
insecure=very should fix it. On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsack gmals...@gmellc.com wrote: Can anyone tell me why my inbound calls keep getting rejected with 401? Here’s the debug information: --- SIP read from UDP:147.135.32.221:5060 --- INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Contact: sip:4144038...@147.135.32.221:5060 Supported: 100rel Max-Forwards: 69 Content-Length: 308 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.248 t=0 0 m=audio 15502 RTP/AVP 0 18 8 96 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 - [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- (11 headers 14 lines) --- [Nov 3 02:08:40] VERBOSE[7207] netsock.c: == Using SIP RTP CoS mark 5 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Sending to 147.135.32.221 : 5060 (NAT) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Using INVITE request as basis request - 31007e...@147.135.32.221 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Found peer 'trunk_1' for '4144038968' from 147.135.32.221:5060 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- Reliably Transmitting (NAT) to 147.135.32.221:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281-;received=147.135.32.221 From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22;tag=as4fffe111 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2dd58be8 Content-Length: 0 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Scheduling destruction of SIP dialog '31007e...@147.135.32.221' in 32000 ms (Method: INVITE) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- SIP read from UDP:147.135.32.221:5060 --- ACK sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 ACK From: Wi Msip:number f...@147.135.32.221;user=phone;tag=9bbc To: usernamesip:s...@216.26.109.22;tag=as4fffe111 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length: 0 Here’s the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register = 6087294351:sip password@sip.broadvoice.com [trunk_1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=6087294351 secret=sip password username=6087294351 insecure=very context=DID_trunk_1 authname=6087294351 dtmfmode=inband dtmf=inband canreinvite=no [guest] type=friend host=dynamic canreinvite=no context=DID_trunk_1 -- _ -- 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
[asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
Hello,
I have this in my dialplan :
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string :
[Nov 3 16:17:27] -- Executing [...@macro-f:43]
Set(SIP/test-0002, vgLabel=vg(1+1)) in new stack
[Nov 3 16:17:27] -- Executing [...@macro-f:44]
Goto(SIP/test-0002, vg(1+1)) in new stack
[Nov 3 16:17:27] NOTICE[23048]: pbx.c:3744 pbx_extension_helper: No
such label 'vg(1+1)' in extension 's' in context 'macro-f'
[Nov 3 16:17:27] WARNING[23048]: pbx.c:9625 pbx_parseable_goto:
Priority 'vg(1+1)' must be a number 0, or valid label
How to overcome this ?!
Asterisk 1.6.2.10
Kind regards,
Jonas.
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Re: [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
On Wed, Nov 3, 2010 at 9:18 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,Set(vgLabel=vg$[${number} + 1])
untested
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Re: [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, November 03, 2010 8:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
Hello,
I have this in my dialplan :
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string :
[Nov 3 16:17:27] -- Executing [...@macro-f:43] Set(SIP/test-0002,
vgLabel=vg(1+1)) in new stack
[Nov 3 16:17:27] -- Executing [...@macro-f:44] Goto(SIP/test-0002,
vg(1+1)) in new stack
[Nov 3 16:17:27] NOTICE[23048]: pbx.c:3744 pbx_extension_helper: No such
label 'vg(1+1)' in extension 's' in context 'macro-f'
[Nov 3 16:17:27] WARNING[23048]: pbx.c:9625 pbx_parseable_goto: Priority
'vg(1+1)' must be a number 0, or valid label
How to overcome this ?!
Asterisk 1.6.2.10
Kind regards,
Jonas.
Don't know about 1.6, but in 1.4 you would do it like this
exten = s,n,Set(vgLabel=$[${number} +1])
This assumes that {number} is a variable and that your not trying to use an
array vg(${number}) or function vg.
For this purpose, I'd use Gotoif instead, although I use a similar concept
in my dialplan, like this
Exten = 1234,1,Goto(foo,${value},1)
[foo]
Exten = s,1,verbose(goto label)
Exten = 1,1,saydigit(1)
Exten = 2,1,saydigit(2)
Exten = 3,1,saydigit(3)
Exten = I,1,playback(invalid-value)
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Re: [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a
string :
Use the MATH function.
Philipp
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Re: [asterisk-users] Issue with asterisk
Hi! It is causing an issue for me. One SIP UA works fine - ring, forward, etc. While the other does not. Make the UAs listen on different ports (for example 5060 and 5062) and see if that solves your problem - if you can't make them have different IPs, that is. Also be sure to fully understand what the insecure= settting in sip.conf does, if you are not then do some reading on that. If you really want to use insecure then you might want to consider to combine that with permit= and deny= to improve on security. You most likely have two SIP UAs that use the same IP, of which the 6839 account is listed last in sip.conf while 3169 is trying to auth (unsuccessfully). The settings that you sent to the list mention neither 6839 nor do they mention 3169. Please state which of the SIP account those are using, on which IP (and port) these phones are. Philipp -- _ -- 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, VoIP and Samsung iDCS100
Hi! 2. Add BRI card(s) to the computer to run Asterisk and somehow hook up the Samsung. Do-able. Connect Asterisk to your ISDN2, then host the Samsung off the asterisk box. But then, might as well dump the Samsung and just put VoIP phones on everyones desks. If you decide to go down this route: You would need a 4-port ISDN card (since 2-port ISDN cards are hard to come by) in order to allow for direct bridging of ISDN-ISDN calls IF you want to operate an anlog fax on the Samsung. direct here means that the call will not travel through the Asterisk core. Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Another way to go is to use an external ISDN gateway (Patton, for example), which will spare you from a lot of kernel driver update headaches in the following years. If you are planning on moving an anlog to an Asterisk-only solution on medium term, then consider to a) keep the Samsung for fax operation, or b) look at an ISDN card that has an option to sync an anlogue telephony card of the same vendor to the clock of the ISDN card (typically using a special but simple card-to-card cable). Philipp -- _ -- 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] Migration from 1.2 to 1.8 in production
Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. Thanks, S. Patel -- _ -- 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, VoIP and Samsung iDCS100
Roger Burton West said at 03/11/2010 12:48: On Wed, Nov 03, 2010 at 12:05:51PM +, Ronny Adsetts wrote: What hardware would I need in the Asterisk so I could hook up some analogue extensions? Am I right in thinking I need something like an FXO/FXS card? Yes, this ought to work. If you're plugging phones into the Samsung it's providing an FXS interface, so you'll need an FXO interface to talk to that; if you want to connect those analogue phones to Asterisk, you'll also need FXS interfaces (though as a short-term fix it would probably be easier to leave them plumbed directly into the Samsung box). Getting four modules (each of which can be FXO or FXS) on a single card is pretty easy (I use an OpenVox A400P from voipon, following recommendations on this list). Cool. The Samsung 8SLI card we have in the Samsung PBX provides 8 analogue ports for regular telephones. I'll plug (some of) these lines from the Samsung in to the Asterisk computer. So an OpenVox 4-FXO card will be on order shortly. You could then connect (some combination of) analogue channels to (some combination of) SIP phones, and vice versa to allow outward dialling. Brill, that's what I hoped. :-) I'll configure Asterisk to route each FXO line direct to a named SIP phone so that dialling 209 for example on the Samsung will go direct to a named SIP account. When a SIP phone does the equivalent of lifting the handset, I'll configure Asterisk to just pass on lifting the handset to its corresponding FXO line. And, assuming I'm not deluded in some way, bingo, problem solved. :-). Once you build the Asterisk-only system, you can use the FXO modules to connect to analogue PSTN lines (assuming you have a use for this). Indeed. Even if not, the OpenVox cards are priced such that it's not a big deal if they don't get reused. Ronny -- Ronny Adsetts Technical Director Amazing Internet Ltd, London t: +44 20 8607 9535 f: +44 20 8607 9536 w: www.amazinginternet.com Registered office: UK House, 82 Heath Road, Twickenham TW1 4BW Registered in England. Company No. 4042957 signature.asc Description: OpenPGP digital signature -- _ -- 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, VoIP and Samsung iDCS100
Philipp von Klitzing said at 03/11/2010 14:10: Hi! Hi :-). 2. Add BRI card(s) to the computer to run Asterisk and somehow hook up the Samsung. Do-able. Connect Asterisk to your ISDN2, then host the Samsung off the asterisk box. But then, might as well dump the Samsung and just put VoIP phones on everyones desks. If you decide to go down this route: You would need a 4-port ISDN card (since 2-port ISDN cards are hard to come by) in order to allow for direct bridging of ISDN-ISDN calls IF you want to operate an anlog fax on the Samsung. direct here means that the call will not travel through the Asterisk core. I think I'm going to skip this route for now - see my other emails. Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Another way to go is to use an external ISDN gateway (Patton, for example), which will spare you from a lot of kernel driver update headaches in the following years. This is good advice anyway for when I build our Asterisk box in the coming months. We'll inevitably stick with ISDN lines for external calls. If you are planning on moving an anlog to an Asterisk-only solution on medium term, then consider to a) keep the Samsung for fax operation, or b) look at an ISDN card that has an option to sync an anlogue telephony card of the same vendor to the clock of the ISDN card (typically using a special but simple card-to-card cable). We currently have our fax line as a DID number on the ISDN/2 which the Samsung routes to an analogue extension which is then answered using Hylafax on a linux box. When we move to a full Asterisk solution I'll potentially just have Asterisk route the fax call via DID number to an FXO/FXS analogue line and have Hylafax deal with it as now. Ronny -- Ronny Adsetts Technical Director Amazing Internet Ltd, London t: +44 20 8607 9535 f: +44 20 8607 9536 w: www.amazinginternet.com Registered office: UK House, 82 Heath Road, Twickenham TW1 4BW Registered in England. Company No. 4042957 signature.asc Description: OpenPGP digital signature -- _ -- 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, VoIP and Samsung iDCS100
How many lines are we talking here? Get a two port T1/PRI Card, use a channel bank, and get your lines from your provider on a PRI. (this way you can start off with 10 numbers, and add up to 300+ and never have to add any extra lines at a per line price. If you looking to save money with SIP providers, you're going to get hit or miss performance with faxing. William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ronny Adsetts Sent: Wednesday, November 03, 2010 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100 Thanks everyone for your replies so far. I've pretty much concluded that going for a full Asterisk solution is the best longer term solution and that's what I'll do. We're moving office before May so that's the perfect time to put in a new phone system. But, I need to implement something quick-time just to buy me some time to do a full Asterisk solution. With that in mind, a couple of questions below: Gordon Henderson said at 02/11/2010 16:39: On Tue, 2 Nov 2010, Ronny Adsetts wrote: 1. Add analogue card(s) to the computer to run Asertisk and treat them as analogue extensions in the Samsung. Statically route each extension to a VoIP handset/user. So incoming via ISDN, Samsung converts to analogue, PC converts to VoIP and then out again - it'll work (maybe), but it's a huge waste of resources. It is a waste of resources I agree but might be the easiest way forward. As I mentioned above, I need a quick and dirty short term solution to buy me some time to scrap the Samsung and replace with Asterisk. I have an analogue extensions card in the Samsung that we currently use for a fax and answer-phone for one of our numbers (long story). What hardware would I need in the Asterisk so I could hook up some analogue extensions? Am I right in thinking I need something like an FXO/FXS card? I imagine I could then route (effectively hard-wire) those extensions direct each to a specific SIP phone using Asterisk? Thanks again for all your help so far. Ronny -- Ronny Adsetts Technical Director Amazing Internet Ltd, London t: +44 20 8607 9535 f: +44 20 8607 9536 w: www.amazinginternet.com Registered office: UK House, 82 Heath Road, Twickenham TW1 4BW Registered in England. Company No. 4042957 -- _ -- 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] Migration from 1.2 to 1.8 in production
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, November 03, 2010 9:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Migration from 1.2 to 1.8 in production Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. Thanks, S. Patel 1.8 will introduce many features and is the supported standard, which will be important to you since you are on a 5 year upgrade plan. It also has more opportunities than the 1.4 version since it is under active development and 1.4 is in a patch only state. If immediate stability is your goal, you may want to stick with 1.4. If I were going to bite the bullet on 1.6, I'd jump straight to 1.8 since there is no end-of-life advantage. -- _ -- 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] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
On Wed, 3 Nov 2010, Philipp von Klitzing wrote: Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Interesting. I've been usng mISDN for some years now without issues. Why should I migrate to DAHDI? Gordon -- _ -- 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] Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing
Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other asterisk systems are using 1.4. I have played around with progressinband in sip.conf with now success. Whatever I set progressinband to, it doesn't seem to change a thing. 183 Session Progress never seems to be called when looking at sip debug. It is only when I use the Progress application before my dial command that I get the 183 Session Progress message in sip debug. We also have a Trixbox system using asterisk 1.6 that had the same problem. The way I fixed that was to set progressinband=yes in sip.conf. This did not work with this system (Yes I know Trixbox is completely different). The only thing that looks different is the order in which 183 Session Progress and 180 Ringing get sent in sip debug. On the troubled Asterisk system 183 gets sent before 180. On the fixed Trixbox system 180 gets sent before 183. Does this mean anything? We also have Polycom Phones which I heard are notorious with ringback issues. Thanks, Chris -- _ -- 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] Migration from 1.2 to 1.8 in production
Am 03.11.10 15:14, schrieb satish patel: Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. Thanks, S. Patel Hello Patel, it hardly depends on how many users and concurrent calls you have in your system cause i have recognized 1.2 can handle much more peers than 1.6 or 1.8. maybe you should try to setup a test server and first try it with your setup and some load tests if everything is working as you expect. best regards stefan -- _ -- 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] all circuits busy now
I am getting all circuits busy now. But with my sip phone I can dial normally this provider... I cant see what's wrong here, any idea how to dig into this? My in/out: canreinvite=yes type=peer qualify=yes insecure=very host= x.x.x.x port=5080 username= xxx secret= xxx context=outbound-allroutes allow=all Thanks in advance. -- Executing [8613430491...@from-internal:1] Macro(SIP/123-00075448, user-callerid|SKIPTTL|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/123-00075448, AMPUSER=123) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/123-00075448, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/123-00075448, 1|Set|REALCALLERIDNUM=123) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/123-00075448, AMPUSER=123) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/123-00075448, AMPUSERCIDNAME=123) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/123-00075448, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/123-00075448, AMPUSERCID=123) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/123-00075448, CALLERID(all)=123 123) in new stack -- Executing [...@macro-user-callerid:9] ExecIf(SIP/123-00075448, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:10] GotoIf(SIP/123-00075448, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [...@macro-user-callerid:19] NoOp(SIP/123-00075448, Using CallerID 123 123) in new stack -- Executing [8613430491...@from-internal:2] Set(SIP/123-00075448, _NODEST=) in new stack -- Executing [8613430491...@from-internal:3] Macro(SIP/123-00075448, record-enable|123|OUT|) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/123-00075448, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/123-00075448, recordingcheck|20101103-174057|1288795257.1108389) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20101103-174057|1288795257.1108389: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/123-00075448, ) in new stack -- Executing [8613430491...@from-internal:4] Macro(SIP/123-00075448, dialout-trunk|25|8613430491011||) in new stack -- Executing [...@macro-dialout-trunk:1] Set(SIP/123-00075448, DIAL_TRUNK=25) in new stack -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/123-00075448, 0?sub-pincheck|s|1) in new stack -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/123-00075448, 0?disabletrunk|1) in new stack -- Executing [...@macro-dialout-trunk:4] Set(SIP/123-00075448, DIAL_NUMBER=8613430491011) in new stack -- Executing [...@macro-dialout-trunk:5] Set(SIP/123-00075448, DIAL_TRUNK_OPTIONS=trf) in new stack -- Executing [...@macro-dialout-trunk:6] Set(SIP/123-00075448, OUTBOUND_GROUP=OUT_25) in new stack -- Executing [...@macro-dialout-trunk:7] GotoIf(SIP/123-00075448, 1?nomax) in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [...@macro-dialout-trunk:9] GotoIf(SIP/123-00075448, 0?skipoutcid) in new stack -- Executing [...@macro-dialout-trunk:10] Set(SIP/123-00075448, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [...@macro-dialout-trunk:11] Macro(SIP/123-00075448, outbound-callerid|25) in new stack -- Executing [...@macro-outbound-callerid:1] ExecIf(SIP/123-00075448, 0|SetCallerPres|) in new stack -- Executing [...@macro-outbound-callerid:2] ExecIf(SIP/123-00075448, 0|Set|REALCALLERIDNUM=123) in new stack -- Executing [...@macro-outbound-callerid:3] GotoIf(SIP/123-00075448, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [...@macro-outbound-callerid:6] Set(SIP/123-00075448, USEROUTCID=) in new stack -- Executing [...@macro-outbound-callerid:7] Set(SIP/123-00075448, EMERGENCYCID=) in new stack -- Executing [...@macro-outbound-callerid:8] Set(SIP/123-00075448, TRUNKOUTCID=) in new stack -- Executing [...@macro-outbound-callerid:9] GotoIf(SIP/123-00075448, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [...@macro-outbound-callerid:12] ExecIf(SIP/123-00075448, 0|Set|CALLERID(all)=) in new stack -- Executing [...@macro-outbound-callerid:13] ExecIf(SIP/123-00075448, 0|Set|CALLERID(all)=) in new stack -- Executing [...@macro-outbound-callerid:14] ExecIf(SIP/123-00075448, 0|SetCallerPres|prohib_passed_screen) in new stack -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/123-00075448, 0|AGI|fixlocalprefix) in new stack -- Executing [...@macro-dialout-trunk:13] Set(SIP/123-00075448, OUTNUM=008613430491011) in new stack -- Executing [...@macro-dialout-trunk:14] Set(SIP/123-00075448, custom=SIP/CHINA01) in new stack -- Executing [...@macro-dialout-trunk:15
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
Thanks for reply, I believe we have around 300 SIP phone register on asterisk and we have 2 T1 line. Roughly i would say max concurrent number 20/30 Max. My only concern is stability after whatever version migration. I believe 1.8 is new and it's just coming out form egg so quite worry about stability. So I have two choice 1.4 and 1.6 stable version. is there anyone who is using 1.8 in production? I am quite impressed with 1.8 features though Thanks, S. Patel Date: Wed, 3 Nov 2010 15:42:21 +0100 From: s...@sil.at To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production Am 03.11.10 15:14, schrieb satish patel: Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. Thanks, S. Patel Hello Patel, it hardly depends on how many users and concurrent calls you have in your system cause i have recognized 1.2 can handle much more peers than 1.6 or 1.8. maybe you should try to setup a test server and first try it with your setup and some load tests if everything is working as you expect. best regards stefan -- _ -- 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] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Hi! Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Interesting. I've been usng mISDN for some years now without issues. Why should I migrate to DAHDI? None - if you are happy then don't touch it. :-) Otherwise search this list's archive. Philipp -- _ -- 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] [SOLVED][BUG??] Asterisk linphone call dropping by itself
Well the problem seems to be: the linphones are listening on port 5062, while * is on port 5060. For some reason, the INVITEs are received from *, but are forwarded on port 5060 by default. I solved the problem by moving * to port 5062 and moving the linphones back to port 5060. All is well, but may this be a bug? Thanks, M Il 03/11/2010 12:48, Matteo Fortini ha scritto: hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two linphones register with a name which is defined as dynamic in sip.conf the call terminates on the caller's side, while the callee is still connected, and I have to force the termination on that side. I'm using asterisk 1.8.0 and linphone 3.99 I really don't know how to investigate further... a capture on sip ports just shows that on the 25th ack packet the other side answers with a BYE instead of with an OK SDP packet. TIA, Matteo -- _ -- 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] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
On Wed, 3 Nov 2010, Philipp von Klitzing wrote: Hi! Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Interesting. I've been usng mISDN for some years now without issues. Why should I migrate to DAHDI? None - if you are happy then don't touch it. :-) Otherwise search this list's archive. Ah, that would be too easy ;-) However I am in the process of upgrading an number of systems from 1.2+mISDN to 1.4 + ... So maybe I'll go and have a look at using DAHDI since I think I've almost got the hand of it for analogue and PRI systems now... So I'll go and do some looking - but you reckon DAHDI and ISDN2e (UK: BRI) is as stable/usable as mISDN might be? Cheers, Gordon -- _ -- 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] Migration from 1.2 to 1.8 in production
I have used 1.4 1.6. I am testing 1.8 for production and it is looking very good. I am making some changes to accommodate some minor dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF issues when used with Sonus on the back end. 1.8 is looking very good and we hope to go production before the end of the year. If you have to change righ now are you using custom dialplan code? If you are I would roll the dice and go for 1.8 this will give you the longest life span. If not there is no real big hit for stepping from 1.4 to 1.8. The other issue is if you want really detailed logging for call records the CEL method in 1.8 is the way to go. You will need to be able to boil the data down but it is there. I have seen a few kinks in the current version but it looks like they will be worked out with some incremental updates. Our hope is to be fully 1.8 on all of our backbone production units by the end of Jan 2011 with our first unit by December 2010. I would shy away of 1.6.x based on our experience. Our 1.6.x boxes will move before our 1.4.x boxes. Thanks Bryant From: Tilghman Lesher tles...@digium.com Sent: Wednesday, November 03, 2010 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: satish patel wrote: We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. 1.8 will introduce many features and is the supported standard, which will be important to you since you are on a 5 year upgrade plan. It also has more opportunities than the 1.4 version since it is under active development and 1.4 is in a patch only state. This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. If immediate stability is your goal, you may want to stick with 1.4. If I were going to bite the bullet on 1.6, I'd jump straight to 1.8 since there is no end-of-life advantage. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
If 1.2 is working fine without any problem then why do you need to upgrade to any newer version? I would suggest don't do it. If you really want to do it just for the sake of doing it, upgrade to 1.4 only, which is the most stable and well tested version of asterisk. Upgrading always causes hickups in the new system, and effects quality of service to the customers. As they say, if its not broken, don't fix it. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-11-03 11:30 AM, Tilghman Lesher tles...@digium.com wrote: On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: satish patel wrote: We are running asterisk 1.2.x version in production environment since ... 1.8 will introduce many features and is the supported standard, which will be important to you... This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. If immediate stability is your goal, you may want to stick with 1.4. If I were going to bite... -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
Its good to know the MATH function because it can do much more and also deal
with floating point numbers. However in your case a simple addition would be
suffice as other posters posted, or try Danny's GotoIf if it fits your
scenario.
Set(vgLabel=vg${MATH(${vg}+1,i)})
Zeeshan A Zakaria
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On 2010-11-03 9:39 AM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,GoTo(${vgLabel})
But in stead of vgL...
Use the MATH function.
Philipp
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Re: [asterisk-users] Migration from 1.2 to 1.8 in production
Thanks a lots Bryant, I would test 1.8 and see if it work out, Definitely 1.8 going to be rock sooner or later, Let's try 1.8 Currently we are facing some issue with echo in conference call with 1.2 version hopefully it will go away with 1.8 Thanks, S. Patel From: brya...@zktech.com To: asterisk-users@lists.digium.com Date: Wed, 3 Nov 2010 11:44:24 -0400 Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production I have used 1.4 1.6. I am testing 1.8 for production and it is looking very good. I am making some changes to accommodate some minor dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF issues when used with Sonus on the back end. 1.8 is looking very good and we hope to go production before the end of the year. If you have to change righ now are you using custom dialplan code? If you are I would roll the dice and go for 1.8 this will give you the longest life span. If not there is no real big hit for stepping from 1.4 to 1.8. The other issue is if you want really detailed logging for call records the CEL method in 1.8 is the way to go. You will need to be able to boil the data down but it is there. I have seen a few kinks in the current version but it looks like they will be worked out with some incremental updates. Our hope is to be fully 1.8 on all of our backbone production units by the end of Jan 2011 with our first unit by December 2010. I would shy away of 1.6.x based on our experience. Our 1.6.x boxes will move before our 1.4.x boxes. Thanks Bryant From: Tilghman Lesher tles...@digium.com Sent: Wednesday, November 03, 2010 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: satish patel wrote: We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. 1.8 will introduce many features and is the supported standard, which will be important to you since you are on a 5 year upgrade plan. It also has more opportunities than the 1.4 version since it is under active development and 1.4 is in a patch only state. This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. If immediate stability is your goal, you may want to stick with 1.4. If I were going to bite the bullet on 1.6, I'd jump straight to 1.8 since there is no end-of-life advantage. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
El 03/11/10 10:44, Bryant Zimmerman escribió: I have used 1.4 1.6. I am testing 1.8 for production and it is looking very good. I am making some changes to accommodate some minor dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF issues when used with Sonus on the back end. 1.8 is looking very good and we hope to go production before the end of the year. If you have to change righ now are you using custom dialplan code? If you are I would roll the dice and go for 1.8 this will give you the longest life span. If not there is no real big hit for stepping from 1.4 to 1.8. The other issue is if you want really detailed logging for call records the CEL method in 1.8 is the way to go. You will need to be able to boil the data down but it is there. I have seen a few kinks in the current version but it looks like they will be worked out with some incremental updates. Our hope is to be fully 1.8 on all of our backbone production units by the end of Jan 2011 with our first unit by December 2010. I would shy away of 1.6.x based on our experience. Our 1.6.x boxes will move before our 1.4.x boxes. Thanks Bryant Hi Bryant, Thanks for sharing your experience, it encourages us to try and test 1.8 throughly before we upgrade our 1.4 boxes, jumping the 1.6 step in the upgrade path. We are also concerned about the short support time that is left for 1.4 and 1.6, so 1.8 would be the best in support terms. This goes in concordance for what I think the asterisk team wants, that is, to focus on only one well supported version instead of having to support several parallel branches which mean more work and cross-fixing between them. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Gotoif changed in 1.8?
Hi Gang,
I'm testing 1.8.0 on one of my machines and this snippet
chokes on line 7 (works fine with 1.4.30)
[tb-account-balance]
exten = s,1,Set(BALCOUNT=0)
exten = s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} ))
exten = s,n(runagi),Set(TEST_RETURN=NONE)
exten =
s,n,AGI(acctbal.agi,${ABA},${digitacc},${digittype},${digitport},${CHANNEL(l
anguage)},${outtype})
exten = s,n,NoOp(Verbose(bal AGI RETURNED ${TEST_RETURN} ))
exten = s,n,Set(BALCOUNT=$[${BALCOUNT} + 1])
exten = s,n,Gotoif($[${BALCOUNT} 3]?tb-account-balance,s,reset_bc)
exten = s,n,Gotoif($[${TEST_RETURN} = OK]?tb-account-balance,s,ok)
exten = s,n,Gotoif($[${TEST_RETURN} = NONE]?tbstart,s,play-main)
exten = s,n,Gotoif($[${TEST_RETURN} = INVACCT]?tbstart,s,readacct)
exten = s,n(invacct),Playback(${invacct})
exten = s,n,Goto(tb-account-balance,s,runagi)
exten = s,n(ok),Set(BALCOUNT=0)
-- Executing [...@tb-account-balance:7] GotoIf(SIP/134-,
0?tb-account-balance,s,reset_bc) in new stack
[Nov 3 14:23:02] WARNING[20937]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
NONE = OK
^
[Nov 3 14:23:02] WARNING[20937]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to doc/tex/channelvariables.tex.
* Executing [...@tb-account-balance:8] GotoIf(SIP/134-,
?tb-account-balance,s,ok) in new stack
Any ideas?
TIA
Danny Nicholas
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Re: [asterisk-users] ADSL Load Balancing
I use the SG560 (http://www.snapgear.com/index.cfm?skey=1557) to do this. Thats perfect. Any idea where they are available? I cant locate a store online. Thanks Dan -- _ -- 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] Gotoif changed in 1.8?
On Wed, Nov 3, 2010 at 4:05 PM, Danny Nicholas da...@debsinc.com wrote:
Hi Gang,
I’m testing 1.8.0 on one of my machines and this snippet
“chokes” on line 7 (works fine with 1.4.30)
[tb-account-balance]
exten = s,1,Set(BALCOUNT=0)
exten = s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} ))
exten = s,n(runagi),Set(TEST_RETURN=NONE)
exten =
s,n,AGI(acctbal.agi,${ABA},${digitacc},${digittype},${digitport},${CHANNEL(language)},${outtype})
exten = s,n,NoOp(Verbose(bal AGI RETURNED ${TEST_RETURN} ))
exten = s,n,Set(BALCOUNT=$[${BALCOUNT} + 1])
exten = s,n,Gotoif($[${BALCOUNT} 3]?tb-account-balance,s,reset_bc)
exten = s,n,Gotoif($[${TEST_RETURN} = OK]?tb-account-balance,s,ok)
exten = s,n,Gotoif($[${TEST_RETURN} = NONE]?tbstart,s,play-main)
exten = s,n,Gotoif($[${TEST_RETURN} = INVACCT]?tbstart,s,readacct)
exten = s,n(invacct),Playback(${invacct})
exten = s,n,Goto(tb-account-balance,s,runagi)
exten = s,n(ok),Set(BALCOUNT=0)
-- Executing [...@tb-account-balance:7] GotoIf(SIP/134-,
0?tb-account-balance,s,reset_bc) in new stack
[Nov 3 14:23:02] WARNING[20937]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
NONE = OK
^
Too many double-quotes?
Try changing line 3 to:
exten = s,n(runagi),Set(TEST_RETURN=NONE)
--
HTH,
-Bob Beers
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Re: [asterisk-users] Gotoif changed in 1.8?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers
Sent: Wednesday, November 03, 2010 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gotoif changed in 1.8?
On Wed, Nov 3, 2010 at 4:05 PM, Danny Nicholas da...@debsinc.com wrote:
Hi Gang,
Im testing 1.8.0 on one of my machines and this snippet
chokes on line 7 (works fine with 1.4.30)
[tb-account-balance]
exten = s,1,Set(BALCOUNT=0)
exten = s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} ))
exten = s,n(runagi),Set(TEST_RETURN=NONE)
exten =
s,n,AGI(acctbal.agi,${ABA},${digitacc},${digittype},${digitport},${CHANNEL(l
anguage)},${outtype})
exten = s,n,NoOp(Verbose(bal AGI RETURNED ${TEST_RETURN} ))
exten = s,n,Set(BALCOUNT=$[${BALCOUNT} + 1])
exten = s,n,Gotoif($[${BALCOUNT} 3]?tb-account-balance,s,reset_bc)
exten = s,n,Gotoif($[${TEST_RETURN} = OK]?tb-account-balance,s,ok)
exten = s,n,Gotoif($[${TEST_RETURN} = NONE]?tbstart,s,play-main)
exten = s,n,Gotoif($[${TEST_RETURN} = INVACCT]?tbstart,s,readacct)
exten = s,n(invacct),Playback(${invacct})
exten = s,n,Goto(tb-account-balance,s,runagi)
exten = s,n(ok),Set(BALCOUNT=0)
-- Executing [...@tb-account-balance:7] GotoIf(SIP/134-,
0?tb-account-balance,s,reset_bc) in new stack
[Nov 3 14:23:02] WARNING[20937]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token',
expecting
$end; Input:
NONE = OK
^
Too many double-quotes?
Try changing line 3 to:
exten = s,n(runagi),Set(TEST_RETURN=NONE)
--
HTH,
-Bob Beers
TAM, Bob! Guess I've got to go through now and unquote my literals...
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Re: [asterisk-users] Gotoif changed in 1.8?
On Wed, Nov 3, 2010 at 4:32 PM, Danny Nicholas da...@debsinc.com wrote:
TAM, Bob! Guess I've got to go through now and unquote my literals...
Hi Danny,
Glad that helped. But on second thought, maybe the better fix is to
remove the double quotes in the Gotoif()'s, like this:
exten = s,n,Gotoif($[${TEST_RETURN} = OK]?tb-account-balance,s,ok)
exten = s,n,Gotoif($[${TEST_RETURN} = NONE]?tbstart,s,play-main)
exten = s,n,Gotoif($[${TEST_RETURN} = INVACCT]?tbstart,s,readacct)
I have noticed that the double-quotes in comparisons are literally included
in the comparison, not 'escaped out', if that is the right expression.
--
-Bob
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Re: [asterisk-users] trixbox - sip trunk with voipwise
You can't do allow= then disallow=all. This will disable all the codec. Try: disallow=all then allow=g729 allow=ulaw On 10-10-29 03:37 AM, Mert Hakk? Bingöl wrote: Hi, No matter I try, I can not register to Voipwise with Trixbox. It is always in unregistered state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com http://sip.voipwise.com fromuser=username host=sip.voipwise.com http://sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer username=username Register string: username:p...@sip.voipwise.com mailto:username%3ap...@sip.voipwise.com Do you have any suggestions? Thank you. Mert -- Jian Gao IT Administrator SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100 -- _ -- 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/Asterisk SCF Project Wiki
For those of you who may have missed the announcements made last week at AstriCon 2010, the Asterisk and Asterisk SCF projects now have a Wiki site available at https://wiki.asterisk.org This site contains a great deal of Asterisk documentation, development plans and other content, with more to come. It also contains Asterisk SCF documentation, development history and much more. The wiki site will allow you to keep track of what is posted/changed there in various ways, including RSS feeds, direct subscription to pages you are interested, and even subscription to entire project spaces. However, if you want to watch an entire project space, it will be more efficient for the wiki site if you instead subscribe to one of the new mailing lists we've setup at lists.digium.com. The asterisk-wiki-changes list will get notified for all content changes and comments posted in the Asterisk project space; the asterisk-scf-wiki-changes list will get the same sort of notifications for the Asterisk SCF project space. -- _ -- 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] doh! chan_dahdi.conf
For those who don't know, (as I just figured out by reading the sourcecode), that all settings for a particular channels must be placed before the channel = entry. Ie, Immediate=no Channel=1-24 Immediate=yes Channel=25-48 Immediate=no Channel=49-72 1-24 will have immediate set to no, 25-48 yes, 49-72 no Maybe someday the config will be [Channels] GlobalOption=Value [1-24] Option=value [25-48] Option=value [49-72] Option=value William Stillwell -- _ -- 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] doh! chan_dahdi.conf
On Wed, Nov 03, 2010 at 06:30:09PM -0400, William Stillwell (Lists) wrote: For those who don't know, (as I just figured out by reading the sourcecode), that all settings for a particular channels must be placed before the channel = entry. Immediate=no Channel=1-24 Immediate=yes Channel=25-48 Immediate=no Channel=49-72 1-24 will have immediate set to no, 25-48 yes, 49-72 no Maybe someday the config will be [Channels] GlobalOption=Value [1-24] Option=value [25-48] Option=value [49-72] Option=value If you use Asterisk = 1.6.1: [port1] ; The name is not used anywhere Option=value dahdichan = 1-24 [port2] dahdichan = 25-48 Option=value [port3] dahdhichan = 49-72 Option=value -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] doh! chan_dahdi.conf
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Wednesday, November 03, 2010 7:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] doh! chan_dahdi.conf On Wed, Nov 03, 2010 at 06:30:09PM -0400, William Stillwell (Lists) wrote: For those who don't know, (as I just figured out by reading the sourcecode), that all settings for a particular channels must be placed before the channel = entry. Immediate=no Channel=1-24 Immediate=yes Channel=25-48 Immediate=no Channel=49-72 1-24 will have immediate set to no, 25-48 yes, 49-72 no Maybe someday the config will be [Channels] GlobalOption=Value [1-24] Option=value [25-48] Option=value [49-72] Option=value If you use Asterisk = 1.6.1: [port1] ; The name is not used anywhere Option=value dahdichan = 1-24 [port2] dahdichan = 25-48 Option=value [port3] dahdhichan = 49-72 Option=value -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- Interesting, im using 1.6.2.13 I tried that but didn't use the dahdichan option. Thanks for the tip. -- _ -- 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] ADSL Load Balancing
On Tue, Nov 2, 2010 at 8:29 PM, Dan Journo d...@keshercommunications.com wrote: Or does this kind of thing need a serious network switch? Why not set MLPPP, assuming your provider supports it. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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
