Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding
I think the order or elements is relevant: [100] disallow=all allow=ulaw allow=g722 or [100] allow=!all,ulaw,g722 should work. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding
Hello Le 15/12/2013 11:07, jg a écrit : I think the order or elements is relevant: [100] disallow=all allow=ulaw allow=g722 or [100] allow=!all,ulaw,g722 should work. [...] Yes, but what about if 100 have g722 as prefered codec? Eg: [100] disallow=all allow=g722 allow=ulaw [101] disallow=all allow=ulaw [rest of the world] disallow=all allow=g722 calls between 100 and 101 should be done using ulaw, all others call in g722 -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding
I see, you do want something like picking g722 provided there is no transcoding. Because Asterisk is a B2BUA it can transcode, so it would choose g722 where the other party is doing g711. For known parties, maybe one could change the SIP configuration on the fly using the Asterisk realtime engine, or modify the settings of the phone with an http request. Generally, an Asterisk configuration option like prioritize_matching_codecs would be needed, but I don't think this is very useful. In this case there should also be all sound files available in g722. Even if you have them, some channels might still be silent as sometimes users choose to get MOH, for example, from the phone itself. Phones usually store sound files in a single format assuming that somebody else is able to transcode if necessary. Please correct me, if my description is incorrect. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 bridge failing
No - but this is a new setup so I can't say it worked before...it just isn't working from the start. I've found the call setup works and once bridged there is one way audio (to the ATA, none from the ATA). And the the connection drops after 30 secs approx because something on the path (or endpoint) realizes something is wrong... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davis [stda...@multiservice.com] Sent: Sunday, December 15, 2013 12:41 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Did you change your network switch recently? Some Digium IAX ATAs do not behave well with Cisco equipment. On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: meant to say restart didn't help either.. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.camailto:mdup...@ocg.ca] Sent: Saturday, December 14, 2013 11:20 PM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Ok just restart -Original Message- From: 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] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [jc...@digium.commailto:jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.comhttp://www.digium.com www.asterisk.orghttp://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 -- _ -- 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 -- Steven Davis VoIP Engineer Multi Service +1-913-663-9748 o +1-913-871-5155 m stda...@multiservice.commailto:stda...@multiservice.com
Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding
On Sun, Dec 15, 2013 at 5:07 AM, jg webaccou...@jgoettgens.de wrote: I think the order or elements is relevant: [100] disallow=all allow=ulaw allow=g722 or [100] allow=!all,ulaw,g722 should work. jg If I choose that order and the phone supports both ulaw and g722 only ulaw will be used. I want to use g722 when available on both devices, fallback to ulaw without transcoding if both devices support it, or transcode if only one device supports ulaw. I looked at the code more and here is what happens. Device 100 dials 101. The sip_new function is called and AST_CODEC_CHOOSE g722 is set as the read/write format. [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7911 sip_new: *** Our native formats are (g722) [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7912 sip_new: *** Joint capabilities are (ulaw|g722) [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7913 sip_new: *** Our capabilities are (ulaw|g722) [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7914 sip_new: *** AST_CODEC_CHOOSE formats are g722 Dial 101 is executed in the dialplan, sip_request_call is called, which in turn calls sip_new. The AST_CODEC_CHOOSE g722 from above becomes the incoming preferred format. We can only have one preferred format as sip_request_call takes in struct ast_format_cap *cap. [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7911 sip_new: *** Our native formats are (ulaw) [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7912 sip_new: *** Joint capabilities are (nothing) [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7913 sip_new: *** Our capabilities are (ulaw) [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7914 sip_new: *** AST_CODEC_CHOOSE formats are ulaw [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7916 sip_new: *** Our preferred formats from the incoming channel are (g722) Asterisk tries to find a common codec between this channels capabilities and the incoming channel preferred format. Of course there are none (g722 and ulaw don't match) so we pick ulaw and transcode. What I am proposing is Asterisk passes fallback formats to sip_request_call. If the joint capabilities are none, then check the fallback formats. In this case it would be ulaw and ulaw. If there is a match switch the incoming channel to that format (ulaw) and AST_CODEC_CHOOSE would be ulaw this for channel. However I'm not sure how to make this change as I don't know my way around the interaction with the Asterisk core and the channels. Ryan -- _ -- 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] Ctrl-W killing entire line, not just last word
I would guess you need to recompile ? On 12 December 2013 20:07, Dotan Cohen dotanco...@gmail.com wrote: On Wed, Dec 11, 2013 at 10:20 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: You need libedit-dev, not libeditline-dev. Thank you Tzafrir. However, even after installing libedit and libedit-dev, Ctrl-W still kills (deletes) to the beginning of the line. -- Dotan Cohen http://gibberish.co.il http://what-is-what.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 -- _ -- 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] Why doesn't Asterisk try to prevent transcoding
On Sun, Dec 15, 2013 at 7:20 AM, jg webaccou...@jgoettgens.de wrote: I see, you do want something like picking g722 provided there is no transcoding. Because Asterisk is a B2BUA it can transcode, so it would choose g722 where the other party is doing g711. For known parties, maybe one could change the SIP configuration on the fly using the Asterisk realtime engine, or modify the settings of the phone with an http request. Generally, an Asterisk configuration option like prioritize_matching_codecs would be needed, but I don't think this is very useful. In this case there should also be all sound files available in g722. Even if you have them, some channels might still be silent as sometimes users choose to get MOH, for example, from the phone itself. Phones usually store sound files in a single format assuming that somebody else is able to transcode if necessary. Please correct me, if my description is incorrect. jg You are correct. Your idea of the prioritize_matching_codecs option is what I am looking for. Yes Asterisk can transcode, but why transcode when you don't need to. If the phone is advertising both formats it should support them. If the phone only supports local MOH in one format then the phone should only advertise that format. If Answer and Playback are called first then the format would have already been sent back in the 200 OK and Asterisk would transcode when Dial is called. If Dial is called first, change the format for the 200 OK and use it for the rest of the call. I haven't looked into what happens with transfers. The idea comes from the following setup. I have 450 users on a FreePBX / Asterisk server with a Sangoma transcoding card. However I am limited in the number of sessions. I also have a number of smaller 10-50 user deployments without transcoding cards. Remote users have phones with g729 Local users have phones with g722,ulaw,g729 SIP Trunks with ulaw,g729 PRIs with ulaw Remote to local should use g729 Local to local should use g722 Remote to SIP trunk should use g729 Local to SIP trunk should use ulaw Local to PRI should use ulaw Remote to PRI would transcode g729 to ulaw If I set these codecs on the devices depending on which side initiates that call transcoding occurs more often than I would like. I could reverse the codec order, however a lower bandwidth codec is chosen in cases where I would prefer a higher bandwidth codec. I looked at this a year ago on Asterisk 1.8 and ended up using ulaw for everything but remote phones. The remote phones end up transcoding g729 to ulaw for most calls. Ryan -- _ -- 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] Why doesn't Asterisk try to prevent transcoding
You are correct. Your idea of the prioritize_matching_codecs option is what I am looking for. Yes Asterisk can transcode, but why transcode when you don't need to. If the phone is advertising both formats it should support them. If the phone only supports local MOH in one format then the phone should only advertise that format. But things may change during a call or you are stuck with whatever the phone uses for its MOH, which only some people use. If Answer and Playback are called first then the format would have already been sent back in the 200 OK and Asterisk would transcode when Dial is called. If Dial is called first, change the format for the 200 OK and use it for the rest of the call. I haven't looked into what happens with transfers. A lot. You need to consider SIP INVITEs as well as Asterisk features (DTMF signals). Some time ago I had a problem with a codec mismatch when only Local channels (Asterisk features uses them) appeared to be involved. The idea comes from the following setup. I have 450 users on a FreePBX / Asterisk server with a Sangoma transcoding card. However I am limited in the number of sessions. I also have a number of smaller 10-50 user deployments without transcoding cards. Is it possible to let the Sangoma card work only on the most demanding codecs? This requires some analysis to estimate the benefits. Another question is whether the user phones are provisioned or not. If provisioned, then you are the maker of rules. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ctrl-W killing entire line, not just last word
On Sun, Dec 15, 2013 at 3:58 PM, Tiago Geada tiago.ge...@gmail.com wrote: I would guess you need to recompile ? I was under the impression that the library was dynamically linked. I am using the Ubuntu binaries for Asterisk, so if someone could confirm that their Asterisk build does in fact kill (delete) a single word on Ctrl-W, then I'll file a bug against the Ubuntu bug tracker. Thanks! -- Dotan Cohen http://gibberish.co.il http://what-is-what.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] Why doesn't Asterisk try to prevent transcoding
On Sun, Dec 15, 2013 at 9:32 AM, jg webaccou...@jgoettgens.de wrote:
Is it possible to let the Sangoma card work only on the most demanding
codecs? This requires some analysis to estimate the benefits. Another
question is whether the user phones are provisioned or not. If provisioned,
then you are the maker of rules.
Most users have both a desk Polycom phone and a soft phone on their mobile
device or laptop. I don't have control over how the soft phones are
provisioned on mobile devices. I've found a workaround that prevents
transcoding for outbound calls.
remote phone
allow=g729
local phone
allow=ulawg729
trunk
allow=ulawg729
In FreePBX extensions_custom.conf I've added the following. This tries to
force the outbound channel to match the inbound channel's format.
[macro-dialout-trunk-predial-hook]
exten =
s,1,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audionativeformat):1:$[${LEN(${CHANNEL(audionativeformat)})}-2]})
Remote to local g729 pass through
Local to remote g729 transcoding
Local to trunk ulaw pass through
Remote to trunk g729 pass through (addressed by the
dialout-trunk-predial-hook)
Trunk to local ulaw pass through
Trunk to remote g729 transcoding
Alternatively I could set trunk allow=g729,ulaw, which would prevent
transcoding for all inbound calls. Outbound from the local phone would use
the hook to change to ulaw.
I still don't have a way to enable the higher quality g722 codec for
internal use without making a transcoding mess. Maybe Asterisk 12 with
pjsip will have a better solution.
Ryan
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Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding
I still don't have a way to enable the higher quality g722 codec for internal use without making a transcoding mess. Maybe Asterisk 12 with pjsip will have a better solution. Currently, I am no longer using g722 anymore for production setups. I had a some SIP-Phone combinations (not Polycom, not Digium) where there were problems with the mean volume when transcoding occured. -- _ -- 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] Why doesn't Asterisk try to prevent transcoding
I have had the issue for years. The problem is that Asterisk developers are removed from the business. We desperately need simple way to eliminate transcoding when unnecessary. Transcoding brings a server to its knees. It is a very simple new setting in sip.conf prioritize_matching_codecs=yes I vote for this new feature. However, I don't have the expertise to write a patch. I would say that only Digium developers could attempt to do this without disrupting the code too much. I also tried to migrate to PJSIP, but had to go back when I realized there was no channel variable contaning the inbound IP address. In general, any channel hast to provide the information to the dialplan, somehow, otherwise we cannot do business. I hope the PJSIP integration matures soon. -- _ -- 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] Why doesn't Asterisk try to prevent transcoding
On 12/15/2013 09:55 PM, CDR wrote: I have had the issue for years. The problem is that Asterisk developers are removed from the business. We desperately need simple way to eliminate transcoding when unnecessary. Transcoding brings a server to its knees. It is a very simple new setting in sip.conf prioritize_matching_codecs=yes Maybe have a look at FreeSWITCH. It's extremely flexible so may offer what you want to do. 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] Why doesn't Asterisk try to prevent transcoding
Yup - its definitely doable in FS. On 15 December 2013 21:18, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 12/15/2013 09:55 PM, CDR wrote: I have had the issue for years. The problem is that Asterisk developers are removed from the business. We desperately need simple way to eliminate transcoding when unnecessary. Transcoding brings a server to its knees. It is a very simple new setting in sip.conf prioritize_matching_codecs=yes Maybe have a look at FreeSWITCH. It's extremely flexible so may offer what you want to do. 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] Ctrl-W killing entire line, not just last word
Always has cleared the entire line.. On 15 December 2013 16:25, Dotan Cohen dotanco...@gmail.com wrote: On Sun, Dec 15, 2013 at 3:58 PM, Tiago Geada tiago.ge...@gmail.com wrote: I would guess you need to recompile ? I was under the impression that the library was dynamically linked. I am using the Ubuntu binaries for Asterisk, so if someone could confirm that their Asterisk build does in fact kill (delete) a single word on Ctrl-W, then I'll file a bug against the Ubuntu bug tracker. Thanks! -- Dotan Cohen http://gibberish.co.il http://what-is-what.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 -- _ -- 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] Multiple IAX2 Trunks Load balancing
Are you looking for something like this?
Note: This will continuously go between the two trunks until the caller
hangs up, can be fixed by adding loop counter.
;
; extensions.conf
;
[LOADBALANCE]
exten = _X.,1,NoOp(Connect to least used trunk)
; - show active count
exten = _X.,n,NoOp(Calls:
${GROUP_COUNT(TRUNK01CNT)}/${GROUP_COUNT(TRUNK02CNT)} of
${MATH(${GROUP_COUNT(TRUNK01CNT)}+${GROUP_COUNT(TRUNK02CNT)},int)})
; - goto least used trunk
exten = _X.,n,GotoIf($[${GROUP_COUNT(TRUNK01CNT)}
${GROUP_COUNT(TRUNK02CNT)}]?TRUNK02,${EXTEN},1:TRUNK01,${EXTEN},1)
;
;
[TRUNK01]
exten = _X.,1,NoOp(Using Trunk 01)
; - set trunk used counter
exten = _X.,n,Set(GROUP()=TRUNK01CNT)
; - dial trunk
exten = _X.,n,Dial(IAX/T01/${EXTEN})
; - add loop counter to stop infinite loop
exten = _X.,n,NoOp(Use next TRUNK02is congestion or chanunavial)
; - next trunk if CONGESTION
exten = _X.,n,GotoIf($[${DIALSTATUS}=CONGESTION]?TRUNK02,${EXTEN},1)
; - next trunk if CHANUNAVAIL
exten = _X.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?TRUNK02,${EXTEN},1)
exten = _X.,n,Hangup()
;
;
[TRUNK02]
; - same as above
exten = _X.,1,NoOp(Using Trunk 02)
exten = _X.,n,Set(GROUP()=TRUNK02CNT)
exten = _X.,n,Dial(IAX/T02/${EXTEN})
exten = _X.,n,NoOp(Use next TRUNK01is congestion or chanunavial)
exten = _X.,n,GotoIf($[${DIALSTATUS}=CONGESTION]?TRUNK01,${EXTEN},1)
exten = _X.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?TRUNK01,${EXTEN},1)
exten = _X.,n,Hangup()
;
; - end
On 12/14/13, 4:41 PM, Muhammad Usman wrote:
Friends let me define the scenario please;
Scenario:
2 asterisk servers (A B) are connected using 05 IAX2 trunks between
them. The machine A is running asterisk Openvpn server in TUN mode
(5 instances with difference IP addresses for clients). The machine B
is running asterisk with 05 OpenVPN clients using 05 bandwidths. The
IAX trunks are established between each pair of P-2-P ip address of
machine A (The OPENVPN Server) machine B (The Openvpn client).
Requirement:
Required dial plan configuration at machine A for incoming calls from
VoIP Switch/VOS which can forward the calls to IAX2 trunks in round
robin fashion like Load Balancing. If any trunk goes down it starts
forwarding the traffic to other available trunks when it gets UP the
dialplan should perform as desired. Like L.B Fail-over scenarios.
On Fri, Dec 13, 2013 at 8:52 PM, Hans Witvliet aster...@a-domani.nl
mailto:aster...@a-domani.nl wrote:
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks.
I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared with other available trunks.
When it gets Up the script is supposed to perform as desired
i.e in
load balance mode.
Thanks in advance.
Hans said:
Perhaps it is possible to do the L.B. at the O.S. or network
level, and let
all trunks appear to asterisk to one single trunk.
Don asks:
What's the value of load balancing multiple IAX trunks between
the same
system pair? What resources are being balanced?
++
Perhaps the O.P. can explain about his intentions...
In some situations it makes sense though:
If you have to connect two servers, and use different kind of
infrastructure / multiple providers...
hw
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(Muhammad ? )
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