[asterisk-users] can Asterisk 16 handle silence suppression from ITSP
Hi: my ITSP use G.711 with VAD and it can not change the settings. I was using Asterisk 13 but the voice quality is not very good. I don't know if asterisk 16 is good enough to handle this kind of situation? or I still need FreeSWITCH in front of Asterisk to get good voice? thanks a lot for help!! Regards, tbskyd -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
2014-09-30 23:52 GMT+08:00 Matthew Jordan mjor...@digium.com: On Sat, Sep 27, 2014 at 10:28 AM, d tbsky tbs...@gmail.com wrote: I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems can set codec, but the document didn't offer any example. i try to use something like PJSIP_MEDIA_OFFER(alaw) but didn't work. can someone give an example for the function? thanks for the help. The function should work on whatever channel it was set on. If you are going to use it on an outbound channel, then you should use a pre-dial handler to apply it to that channel. it sounds good. could you give out an one line dialplan example so I can try to use it? and the real thing I want to change is the inbound codec, can it work like the chan_sip channel variable SIP_CODEC_INBOUND? thanks a lot for your help!! Regards, tbskyd -- _ -- 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] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
2014-09-28 14:01 GMT+08:00 Markus unive...@truemetal.org: Am 27.09.2014 17:28, schrieb d tbsky: can someone give an example for the function? thanks for the help. Not a programmer here, just grep -r'ed through the code, but maybe try one of these: G711A G711_ALAW thanks a lot for help!! I tried both but none works. maybe this function can not work like the old channel variable SIP_CODEC, which can change inbound call codec. but I do notice something different between chan_sip and chan_pjsip. I use zoiper softphone for testing: when I dialout sip trunk with chan_sip, the remote peer rings, and zoiper now shows what codec to use. if I use SIP_CODEC before dial to change the codec, zoiper will use the new CODEC, but asterisk internal won't change and still transcoding in the middle.(at least core show channel sip/x told me transcoding) when I dialout sip trunk with chan_pjsip, the remote peer rings, but zoiper didn't show what codec to use. only after the callee answer the phone, zoiper shows what codec to use. so it seems chan_pjsip have better chance to do the right thing without transcoding. it's sad that chan_pjsip won't select best codec match two peers automatically without transcoding. but I hope it at least can provide a magic function or channel variable like SIP_CODEC/SIP_CODEC_INBOUND to make correct codec selection. Regards, tbskyd -- _ -- 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] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didn't work with pjsip in asterisk 12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems can set codec, but the document didn't offer any example. i try to use something like PJSIP_MEDIA_OFFER(alaw) but didn't work. can someone give an example for the function? thanks for the help. Regards, tbskyd -- _ -- 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] Change codec when dial from SIP to DAHDI
2014-09-25 20:46 GMT+08:00 Matthew Jordan mjor...@digium.com: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite That article is in the development section of the wiki. While that doesn't mean any of the information there is necessarily wrong, its purpose was to coordinate development efforts, not to define behavior for end-users. In this particular case, portions of that page only affect chan_pjsip: thanks a lot for the hint! you really save my day! I was thinking about studying freeswitch, since people said freeswitch can do that without transcode. now i will spent my time to study chan_pjsip, and hope it can fix the problem. i really want to stay with asterisk :) thanks again for your kindly help!! Regards, tbskyd -- _ -- 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] Change codec when dial from SIP to DAHDI
2014-09-26 3:34 GMT+08:00 Eric Wieling ewiel...@nyigc.com: You will find not transcoding much less useful that one might imagine. hi: can you give some more hint about the topic? in my testing, if the sip phone use G.722 and the sip trunk use G.711, I can hear the quality is not as good as both side use G.711. but you maybe right when both legs use G.711 but transcoding in the middle. the quality seems not so bad but I have test it very deeper.. thanks a lot for your help!! Regards, tbskyd -- _ -- 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] Change codec when dial from SIP to DAHDI
hi:
forgot to mention. not only dialout DAHDI, even I dialout SIP
TRUNK, the situation is the same:
asterisk transcode in the middle even two legs use the same code.
2014-09-25 11:20 GMT+08:00 d tbsky tbs...@gmail.com:
hi:
I Have tried asterisk 1.6.2, 1.8, 11, 12, 13. all versions behave
the same = transcode in the middle even two legs use the same code.
but I found an article which seems to solve this kind of problem:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
but I tried version 13 and didn't notice the change, are there new
diaplan commands or channel variables to do this?
thanks a lot for help!!
Regards,
tbskyd
2014-09-24 1:30 GMT+08:00 d tbsky tbs...@gmail.com:
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec when dial from SIP to DAHDI. I tried
to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP codec
change after dahdi answered the channel. so everything is broken. the
call log like below:
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk-predial-hook:2]
Set(SIP/222-0004, SIP_CODEC=alaw) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk-predial-hook:3]
Set(SIP/222-0004, SIP_OUT_CODEC=alaw) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk-predial-hook:4]
MacroExit(SIP/222-0004, ) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:18] GotoIf(SIP/222-0004,
0?bypass,1) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:19] ExecIf(SIP/222-0004,
1?Set(CONNECTEDLINE(num,i)=0912345678)) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:20] ExecIf(SIP/222-0004,
1?Set(CONNECTEDLINE(name,i)=CID:222)) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:21] GotoIf(SIP/222-0004,
0?customtrunk) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:22] Dial(SIP/222-0004,
DAHDI/g2/0912345678,300,Tt) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] app_dial.c: --
Called DAHDI/g2/0912345678
[2014-09-23 21:18:53] VERBOSE[11634][C-000d] app_dial.c: --
DAHDI/2-1 answered SIP/222-0004
[2014-09-23 21:18:53] NOTICE[11634][C-000d] chan_sip.c: Changing
codec to 'alaw' for this call because of ${SIP_CODEC} variable
[2014-09-23 21:18:53] NOTICE[11634][C-000d] chan_sip.c: Changing
codec to 'alaw' for this call because of ${SIP_CODEC} variable
if I check channel with core show channel x, got DAHDI/SIP
legs final like this:
NativeFormats: (alaw)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: Yes (slin)-(alaw)
ReadTranscode: Yes (alaw)-(slin)
although two legs finally use alaw both, but transcode use slin in
the middle. is it possible to prevent the transcode?
if that is not possible, then maybe I should give up using G722 as
the preffered codec of ip phone. back to G711 seems much easier to
make all legs with the same codec.
thanks a lot for help!!
Regards,
tbskyd
--
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[asterisk-users] Change codec when dial from SIP to DAHDI
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec when dial from SIP to DAHDI. I tried
to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP codec
change after dahdi answered the channel. so everything is broken. the
call log like below:
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk-predial-hook:2]
Set(SIP/222-0004, SIP_CODEC=alaw) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk-predial-hook:3]
Set(SIP/222-0004, SIP_OUT_CODEC=alaw) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk-predial-hook:4]
MacroExit(SIP/222-0004, ) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:18] GotoIf(SIP/222-0004,
0?bypass,1) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:19] ExecIf(SIP/222-0004,
1?Set(CONNECTEDLINE(num,i)=0912345678)) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:20] ExecIf(SIP/222-0004,
1?Set(CONNECTEDLINE(name,i)=CID:222)) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:21] GotoIf(SIP/222-0004,
0?customtrunk) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] pbx.c: --
Executing [s@macro-dialout-trunk:22] Dial(SIP/222-0004,
DAHDI/g2/0912345678,300,Tt) in new stack
[2014-09-23 21:18:46] VERBOSE[11634][C-000d] app_dial.c: --
Called DAHDI/g2/0912345678
[2014-09-23 21:18:53] VERBOSE[11634][C-000d] app_dial.c: --
DAHDI/2-1 answered SIP/222-0004
[2014-09-23 21:18:53] NOTICE[11634][C-000d] chan_sip.c: Changing
codec to 'alaw' for this call because of ${SIP_CODEC} variable
[2014-09-23 21:18:53] NOTICE[11634][C-000d] chan_sip.c: Changing
codec to 'alaw' for this call because of ${SIP_CODEC} variable
if I check channel with core show channel x, got DAHDI/SIP
legs final like this:
NativeFormats: (alaw)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: Yes (slin)-(alaw)
ReadTranscode: Yes (alaw)-(slin)
although two legs finally use alaw both, but transcode use slin in
the middle. is it possible to prevent the transcode?
if that is not possible, then maybe I should give up using G722 as
the preffered codec of ip phone. back to G711 seems much easier to
make all legs with the same codec.
thanks a lot for help!!
Regards,
tbskyd
--
_
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Re: [asterisk-users] skype for asterisk usage in the future
hi: thanks for all the information. I don't use skype and I ban skype at our network. but there are some people who use skype and want us to use skype to contact them. SFA is my saver because our users can use their phone to talk with skype users and no need to install any skype software. I hope skype die asap. but if it is alive, I must find someway to satisfy these skype customers... 2011/7/12 A J Stiles asterisk_l...@earthshod.co.uk: On Tuesday 12 Jul 2011, d tbsky wrote: hi: I am a SFA (skype for asterisk) user. I had ask Digium questions about SFA usage in the future. but they seem too busy to reply. so I tried at this list. I hope there are SFA users or Digium people can solve my confusion. Poor you! To my mind, Skype with its opaque, proprietary protocols is the exact opposite of what telecommunications is supposed to be about. I can make a call, or send an SMS, from my HTC on Vodafone to my friend's Samsung on Tesco without thinking twice about it, and that's the way we all expect it to be. But if it hadn't been for governments enforcing standards, the mobile networks could well have ended up fragmentated; with different handset manufacturers and different network operators all using competing, proprietary standards to lock one another out and their customers in. 2. I saw SFA will not be supported after two years. my question is: although it is not supported, can I still use it? I want to buy more licenses now if I can still use it after two years even without official support. That depends entirely on whether Skype update their protocols and block out the old ones. Two years is easily long enough for them to do that; especially given the way Skype works. It could even do stealth upgrades in the background. Look at it this way: You've got two years to migrate away from Skype and start using something else -- and this time around, for the love of all that's sane and wholesome, be sure to choose something that supports open standards, so you can never get shafted the same way again. If someone manages successfully to reverse-engineer Skype during that time, you *might* have a little longer; but I wouldn't bet the family farm on that. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] skype for asterisk usage in the future
hi: I am a SFA (skype for asterisk) user. I had ask Digium questions about SFA usage in the future. but they seem too busy to reply. so I tried at this list. I hope there are SFA users or Digium people can solve my confusion. 1. SFA can not be registered after 26 July. so I want to prepare a backup machine for our server. I read in the document that I can re-register my SFA once. so I want to make sure if I can re-register with my backup server now, and in the same time my production machine still function correctly. and if my production machine is broken one day, my backup machine can go on line. in short: can I keep two machines with one license now? I hope so because I can not re-register later after 26 July. 2. I saw SFA will not be supported after two years. my question is: although it is not supported, can I still use it? I want to buy more licenses now if I can still use it after two years even without official support. thanks a lot for your kindly help!! Regards, tbskyd -- _ -- 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] 1.8 and prematuremedia problem
hi: maybe you can try noload res_timing_timerfd in modules.conf and see what asterisk pick up for timing. in my system, if I disable res_timing_timerfd, then dahdi timing is selected and system become stable. Regards, tbskyd 2011/5/14 satish patel satish...@hotmail.com: You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card. _S From: tu...@canistec.com Date: Fri, 13 May 2011 18:30:52 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.so DAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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] 1.8 and prematuremedia problem
hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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] 1.8 and prematuremedia problem
hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is res_timing_timerfd module stable in 1.8?
hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using res_timing_dahdi or I can use res_timing_timerfd to get some benefit if I upgrade to 1.8? thank a lot for information!! Regards, tbskyd -- _ -- 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] fsk callerid with DTAS start(like dtmf in issue 9096)?
hi: our country use ETSI Standard ETS 300 659-1 to send caller-id. the caller-id format may be DTMF or FSK. with the newest patch in issue 9096 (9096-1.6.2-branch.diff), the DTMF part is solved now. but asterisk still can not detect the FSK part. in the standard the FSK will send with a DT-AS (dual tone alerting signal) start. the method is described in pdf below(page 12): http://www.araxinfo.com/~bacvic/ets_30065901.pdf I wonder if asterisk can treat this kind of FSK just like the DTMF in issue 9096. issue 9096 add option cidstart=dtmf to chan_dahdi.conf. can we add cidstart=dtas or cidstart=fsk to detect FSK also? is there a easy way to patch FSK like the DTMF? I may need to spent time to study the patch. I hope someone can give me some advice or direction. thanks a lot for help!! Regards, tbskyd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fsk callerid with DTAS start?
hi: in our country callerid is sent with fsk. but it will sent DTAS(dual tone alerting signal) first, then fsk callerid, then first ring. I search google, but didn't find the configuration method or patch for this. any good suggestion for asterisk to detect this? I have trid asterisk 1.4 and 1.6.2. thanks for help!! Regards, tbskyd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM on Do Not Call list????
Hi: yes. that is very interesting. i hope they can resolve that and back to work soon. i have many technical issues need their support. i have heard many positive reports about snom. in my testing experience, their product seems a little better than grandstream. snom has more functions and looks good. but grandstream bt200 is much more stable. (didn't try gxp2000, too many bad reports). their forum(http://forum.snom.com) seems offine too now. i hope they can hire some people to handle these things. Regards, tbskyd 2008/3/13, Drew Gibson [EMAIL PROTECTED]: Some light relief SNOM say Please note that you will not be able to reach us by phone. http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] snom official english forum
Hi: snom now has an official english forum for their products: http://forum.snom.com/ there seems few people know the place so there are only a few posts. hope snom experts can take a look and maybe leave some experience for others.. Regards, tbskyd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi: i have discussed the transfer function with grandstream engineers. their operation procedure is complicated(eg: no attended+blind transfer). i tell them the simple way, but got no response since then. Regards, tbskyd 2008/1/7, Andrew Joakimsen [EMAIL PROTECTED]: Another thing is I've found the grandstream phone way of doing things like transfer, etc much easier to understand for laypeople than the more expensive phones. There is no clutter of keys or menus. On Dec 23, 2007 11:50 AM, d tbsky [EMAIL PROTECTED] wrote: hi: thanks a lot for so many great information. i tried to read the specs and manuals for all the phones mentioned. we use alcatel pbx in most offices. i surveyed some users to understand what functions they use most. and i found few people know how to use 3way-conf or forward.i think if the function needs two or more keys to operate, then people tend to ignore it unless he use that function for daily business. i conclude the functions we need are all basic functions. but due to the difference of ip pbx/phones and classic pbx/phones, some of these functions seem not so basic in the ip world: 1. dial out name display. when you dial a number, the phone lcd will show the corresponding name, so you can realize if it is the correct number immediately. this needs a corporate directory support, or put the whole corporate phonebooks to every ip phone. most ip phone has less than 500 local phonebook entries. this is not enough for us. grandstream: has xml phonebook support and can combine with local phonebooks. linksys: has coporate directory but seems only work with linksys pbx, not asterisk. aastra: has xml phonebook snom: has ldap and xml phonebook. xml seems for browsing,don't know if work here. other china brand phone: none. 2. transfer. transfer is simple and straightforward in classic pbx. you just press transfer then dial number and you are on the way of attended transfer. you press transfer again to cancel transfer. you hangup to complete the attended transfer. if you hangup before the completion of attended transfer, the transfer becomes blind transfer automatically. eventually user didn't notice the blind or attended concept in classic pbx. snom: has transfer on hook. don't know if it can do all what i want. others: some china phones almost can do it, but need to press hold to cancel transfer. 3. call back on busy. in alcatel, if you dial someone and he is on the phone, you will hear something like busy, please dial 5 if you want to request callback. you can dial 5 and you will hear success, please hangup. asterisk has several ways and patches to do this. but i saw some phone can do this locally. i don't know which is better. linksys: has this function in spec. don't know how to use. snom: has call completion. others: i didn't find this or i miss it. 4. pickup. i think this is easy to emulate *8 and let asterisk do it. any better method? every phones can do this emulation. 5. three-way conference, forward. if there are simple (one key) method to implement these. in alcatel, if the phone if forwarded, when you pick up the handset you will hear like forwarded, please press *1 to cancel. it's easy so everyone can cancel the forward. but it need two keys to start a forward, so few users know how to forward a number. please correct me if there are mistakes or missing. thanks again for your great help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
Hi lan: thanks for your reply. i already discussed with atcom engineer. they are sorry that they can not satisfy any of my request. they will release an advanced model this year and hope it can catch up others. fanvil is really poor. we have dozens of fanvil FV6050 and now we have to give up all of them and waiting for snom phones. Regards, tbskyd 2008/1/4, Ian FREISLICH [EMAIL PROTECTED]: d tbsky wrote: hi: thanks for the information. you are the second one who mentioned atcom. so i think this phone has basic quality. i don't have atcom in hand. but i have other china brand(fanvil) phone which seems the same as atcom: infeneon based, sip, iax, good sound quality. but it has poor firmware support and limited function. i check the atcom manual, but didn't find the functions i need (corporate phonebook, transfer, callback..etc). I think that Fanvil is ATCom repackaged (I have an atcom and fanvil phone and the configuration structure and menus are the same although the atcom interface looks better). Fanvil's firmware support is poor and I accidentally downgraded the firmware thinking I was upgrading it according to their web page. Phonebook will be an issue. Attended an unattended transfers aren't a problem with these phones and I thought call-back would be implimented in the PBX, not the phone. Maybe have a look at Mitel, you can tickle a URL on the phone to make it dial, so clicking on the name on the company directory in the intranet will call them using your phone. Ian -- Ian Freislich ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi: thanks a lot for so many great information. i tried to read the specs and manuals for all the phones mentioned. we use alcatel pbx in most offices. i surveyed some users to understand what functions they use most. and i found few people know how to use 3way-conf or forward.i think if the function needs two or more keys to operate, then people tend to ignore it unless he use that function for daily business. i conclude the functions we need are all basic functions. but due to the difference of ip pbx/phones and classic pbx/phones, some of these functions seem not so basic in the ip world: 1. dial out name display. when you dial a number, the phone lcd will show the corresponding name, so you can realize if it is the correct number immediately. this needs a corporate directory support, or put the whole corporate phonebooks to every ip phone. most ip phone has less than 500 local phonebook entries. this is not enough for us. grandstream: has xml phonebook support and can combine with local phonebooks. linksys: has coporate directory but seems only work with linksys pbx, not asterisk. aastra: has xml phonebook snom: has ldap and xml phonebook. xml seems for browsing,don't know if work here. other china brand phone: none. 2. transfer. transfer is simple and straightforward in classic pbx. you just press transfer then dial number and you are on the way of attended transfer. you press transfer again to cancel transfer. you hangup to complete the attended transfer. if you hangup before the completion of attended transfer, the transfer becomes blind transfer automatically. eventually user didn't notice the blind or attended concept in classic pbx. snom: has transfer on hook. don't know if it can do all what i want. others: some china phones almost can do it, but need to press hold to cancel transfer. 3. call back on busy. in alcatel, if you dial someone and he is on the phone, you will hear something like busy, please dial 5 if you want to request callback. you can dial 5 and you will hear success, please hangup. asterisk has several ways and patches to do this. but i saw some phone can do this locally. i don't know which is better. linksys: has this function in spec. don't know how to use. snom: has call completion. others: i didn't find this or i miss it. 4. pickup. i think this is easy to emulate *8 and let asterisk do it. any better method? every phones can do this emulation. 5. three-way conference, forward. if there are simple (one key) method to implement these. in alcatel, if the phone if forwarded, when you pick up the handset you will hear like forwarded, please press *1 to cancel. it's easy so everyone can cancel the forward. but it need two keys to start a forward, so few users know how to forward a number. please correct me if there are mistakes or missing. thanks again for your great help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi: thanks for the information. you are the second one who mentioned atcom. so i think this phone has basic quality. i don't have atcom in hand. but i have other china brand(fanvil) phone which seems the same as atcom: infeneon based, sip, iax, good sound quality. but it has poor firmware support and limited function. i check the atcom manual, but didn't find the functions i need (corporate phonebook, transfer, callback..etc). Regards, tbskyd 2007/12/24, Vidura Senadeera [EMAIL PROTECTED]: Hi, Try atcom. www.atcom.com.cn We have tested atcom and its quality also good. they are using infeneon chipset. its support asterisk, sip, iax as well.decent look. cost effetive. still they have basic ip phone modles. starting from next year they will release new modles. Regards, vidura. -- Message: 1 Date: Fri, 21 Dec 2007 12:04:57 +0100 From: Fredrik S?derlund [EMAIL PROTECTED] Subject: Re: [asterisk-users] ip phone suggestion for Asia? To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Check out yntx www.yntx.com fear prices and recides in Asia and iss it sip on asteriks they will do ! try to buy one to trye it out before buying fore hole company.. /MVH Fille ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi Fredrik : thanks for your information. after checking yntx manuals, i found i have one phone in my hand, which has the same firmware with yntx phones. although it is a different brand and looks different. the phone's basic function is ok, but we need some advanced functions like xml phonebook. i hope these china phones would catch up quickly so we all can have better, cheaper phones. Regards, tbskyd 2007/12/21, Fredrik Söderlund [EMAIL PROTECTED]: Check out yntx www.yntx.com fear prices and recides in Asia and iss it sip on asteriks they will do ! try to buy one to trye it out before buying fore hole company.. /MVH Fille ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi mkezys: ok. i will add linksys to our testing list. but cisco tend to lock things. can we get firmware for linksys easily ? or we must pay like cisco routers and switches? 2007/12/21, Mindaugas Kezys [EMAIL PROTECTED]: Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive. We use them (SPA942) in our company. Everybody's happy. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky Sent: Thursday, December 20, 2007 6:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ip phone suggestion for Asia? Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] put fxo channel before E1 channel?
hi: my system has one 4-port fxo card and one 2-port E1 card. for some reason, i like to place fxo as channel 1-4, and E1 use the rest channels (5-66). i modify zaptel.conf, and ztcfg -vv is happy. but asterisk seems not happy with this configuration. it still want channel 16 as D-channel, in my case the D-channel should be 20. i don't know if this is a limit of asterisk. i play some parameters in zapata.conf like trunkgroup. but i still can not get it work. any suggestion? or it is not allowed? thanks for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi gnubie: snom seems has some re-brand ip phones. do they use the same firmware? if they are the same, i don't understand why snom do this.. Regards, tbskyd 2007/12/20, GNUbie [EMAIL PROTECTED]: On Dec 20, 2007 12:33 PM, d tbsky [EMAIL PROTECTED] wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Try getting the Aztech IP150 http://www.aztech.com.sg/ip_telephony/ip150.html which is based on the SNOM 300. Regards, GNUbie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] put fxo channel before E1 channel?
hi jsmith: that explains everything. i didn't aware the module load sequence would cause big difference. is there any document about this i am missing? now the system is working as expected. i m glad that i asked and you answered. thanks a lot for your quick reply and help!! Regards, tbskyd 2007/12/21, Jared Smith [EMAIL PROTECTED]: On Thu, 2007-12-20 at 23:39 +0800, d tbsky wrote: hi: my system has one 4-port fxo card and one 2-port E1 card. for some reason, i like to place fxo as channel 1-4, and E1 use the rest channels (5-66). This will only work if you load the kernel driver for the fxo card before the kernel driver for the E1 card. To see which order they've come up in, you can check the /proc/zaptel directory. You should see a file in that directory for each span, and if you look at the contents of each file, you'll see which channels are in that span. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ip phone suggestion for Asia?
Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi Joel: thanks a lot for your reply. i forgot snom :) i wrote a snom employ found in this email list, but got no reply. i saw there are huge complain about grandstream firmware this year. grandstream seems response and solve some of them. i wonder if their product is ok now. i don't know the situation of snom, will they response to user's request? thanks again for your kindly help!! Regards, tbskyd 2007/12/20, Joel Hill [EMAIL PROTECTED]: Hi tbskyd, We have found that the Grandstream's are not that great a phone. One of our best sellers is the Snom range and I know that the Australian supplier spends half his time in Hong Kong so you shouldn't have any problems getting so over there. They are a little more expensive than the Grandstream's but cheaper than the Polycoms around that Aastra price range. Cheers, Joel. On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi Joe: thanks for your great opinion. we will buy spares of course. our new office needs about 150 phone.we use alcatel before. but want to change to asterisk now for better future. since aastra is not sale to asia (that's sad, aastra boss seems born at taiwan). i will choose snom or grandstream. polycom seems really expensive, i saw someone say their cheap model has no hardware transfer key. so i think i need to pay much more to get what i want in polycom. Regards, tbskyd 2007/12/20, Joe [EMAIL PROTECTED]: You get what you pay for. Snoms are good phones. Grandstreams are also good. I hear Snoms are easier to get around NAT and the seem like higher quality construction. Grandstreams are great for cheap and easy set-ups. I remember one guy telling me he buy up a case and if anything goes wrong with a unit he has a couple spares to go in its place. Over all... if you're looking for setting up office phones, Snom, Polycom, and Aastra look/feel nice. If you're looking to set up small offices or call centers on the cheap, Grandstreams are OK. I personally like Grandstream for home use, but I use Polycom for work, and I hear good things about Snom. On Dec 20, 2007 12:06 AM, d tbsky [EMAIL PROTECTED] wrote: hi Joel: thanks a lot for your reply. i forgot snom :) i wrote a snom employ found in this email list, but got no reply. i saw there are huge complain about grandstream firmware this year. grandstream seems response and solve some of them. i wonder if their product is ok now. i don't know the situation of snom, will they response to user's request? thanks again for your kindly help!! Regards, tbskyd 2007/12/20, Joel Hill [EMAIL PROTECTED]: Hi tbskyd, We have found that the Grandstream's are not that great a phone. One of our best sellers is the Snom range and I know that the Australian supplier spends half his time in Hong Kong so you shouldn't have any problems getting so over there. They are a little more expensive than the Grandstream's but cheaper than the Polycoms around that Aastra price range. Cheers, Joel. On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
i just met the same problem. i want to match extension that end with a number, but can not find a way. i also found that _.X match all extension, but won't match any caller-id number in dialplan. maybe it is a bug. but it seems not important since _.X is useless anyway. 2007/9/15, Tilghman Lesher [EMAIL PROTECTED]: On Friday 14 September 2007 11:39:40 Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) The problem you're seeing is that the period is a short-circuit operator. It says if you match everything so far and at least one more character, then you have a match, no need to go any further. You CANNOT match past a '.'. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what is the usable feature in DUNDi?
hi: i create a dundi environment by the caveman can do it dundi guide. it works fine.but i want to extend the example for my own need, so i follow the sample dundi.conf config file comes with asterisk 1.4.11 source. i try to use precache and failed, and there seems no one know how to use it after googling. i try to setup dynamic dundi with [*] and failed, and google tell me that feature is not implement yet. there is a patch to fix this: http://bugs.digium.com/view.php?id=10546nbn=1 i can only find one kind of syntax example about the mappings keyword. there seems no other way about how to using mappings. so i wonder what's the situation of DUNDi today. what's the usable feature in asterisk 1.4? will it become mature in the 1.4 release? or we should wait for 1.6? thanks for help!! Regards, tbskyd ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to DUNDi branch office with area code?
hi: thanks a lot for your suggestion. i have setup up an experimental environment like yours, and dundi works great. but it's not easy for us to archive this in real world. we have other pbx(like alcatel) that need to co-work with asterisk. so area code with each branch office seems easier to maintain. hope we can run pure asterisk one day.. 2007/9/8, Bruce Reeves [EMAIL PROTECTED]: DUNDi can be used in branch office and I have a similar setup to what your are referring with 11 sites. One thing that I decided to do, but did not have to is define site extensions like you did, but I use the 4 digit extension locally and via dundi. Here is the details: In my case I check for the 4 digit extension in the current site then do a look up on dundi. Each site send the request to 2 core systems that keep up with all the peers and forward the request on.Once the extension is found in the dundi context on a server the call gets routed to the correct site.By doing it this way I don't have to remember to dial a specific number for an intra site call. The other thing is I do I have each site a set of numbers, like 3100 - 3199 is site B and 3200 - 3299 is C, but with DUNDi I do not have to do it that way, it will find the extensions since I use regexten=whatever extension in my sip.conf for each phone. I hope that makes sense, JR has done an excellent job explaining DUNDi in several white papers, and I have used something from all of them. On 9/6/07, d tbsky [EMAIL PROTECTED] wrote: hi: i am new to asterisk and dundi. we have some branch office which will use asterisk in the future. they will form a full-mesh structure so every site can contact each other directly. i want to try setup dundi, then we don't need to modify every pbx when a new site add in the cloud. thanks to the great dundi document caveman can do it and other resource in the voip-info.org. i learn the basic setup of dundi. but i want to a little advanced setup with area code. like this: site HQ: has extension 101,102,103, and site HQ has area code 99 site A: has extension 101,102,103, and site A has area code 01 site B: has extension 101,102,103 and site B has area code 02 site C: has extension 101,102,103 and site C has area code 03 we want to use 4 as prefix to call to the internal cloud. so user at site A can call 4-99-101 to contact extension 101 at HQ. site B can call 4-03-102 to contact extension 102 at site C. now i m confused about this structure with DUNDi. i don't know the best way to setup DUNDi for this structure. i think maybe i should do below when user call 4-99-101 at site A : 1. site A ask for dundi request 4-99-101 to site HQ 2. site HQ strip 4-99 and look up 101 at local context 3. site HQ return the destination to site A 4. site A use the destination to call extension 101 at site HQ i don't know if step 23 is possible in dundi.conf. the example in the internet didn't tell how to do this. or there are better/standard ways to do this? thanks a lot for any suggestion!! Regards, tbskyd ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to DUNDi branch office with area code?
hi: i am new to asterisk and dundi. we have some branch office which will use asterisk in the future. they will form a full-mesh structure so every site can contact each other directly. i want to try setup dundi, then we don't need to modify every pbx when a new site add in the cloud. thanks to the great dundi document caveman can do it and other resource in the voip-info.org. i learn the basic setup of dundi. but i want to a little advanced setup with area code. like this: site HQ: has extension 101,102,103, and site HQ has area code 99 site A: has extension 101,102,103, and site A has area code 01 site B: has extension 101,102,103 and site B has area code 02 site C: has extension 101,102,103 and site C has area code 03 we want to use 4 as prefix to call to the internal cloud. so user at site A can call 4-99-101 to contact extension 101 at HQ. site B can call 4-03-102 to contact extension 102 at site C. now i m confused about this structure with DUNDi. i don't know the best way to setup DUNDi for this structure. i think maybe i should do below when user call 4-99-101 at site A : 1. site A ask for dundi request 4-99-101 to site HQ 2. site HQ strip 4-99 and look up 101 at local context 3. site HQ return the destination to site A 4. site A use the destination to call extension 101 at site HQ i don't know if step 23 is possible in dundi.conf. the example in the internet didn't tell how to do this. or there are better/standard ways to do this? thanks a lot for any suggestion!! Regards, tbskyd ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
