Re: [asterisk-users] leading ghost 0
Hi Leandro, I cannot restart dahdi because the PBX is in production, all I can do is a module reload chan_dahdi.so. Giorgio On 11/20/2012 03:52 PM, Danny Nicholas wrote: In my past experience the best recourse for dealing with a DAHDI trunked asterisk system is this sequence Service asterisk stop Service dahdi restart Service asterisk start *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Leandro Dardini *Sent:* Tuesday, November 20, 2012 8:50 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] leading ghost 0 Not only, you have to restart dahdi/zaptel as well. Leandro 2012/11/20 Frederic Van Espen frederic...@gmail.com mailto:frederic...@gmail.com On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote: I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Somebody correct me if I'm wrong but I think you have to restart asterisk when you change these settings on dahdi. Keep that in mind. Cheers, Frederic -- _ -- 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] Simultaneous caller/callee hangup; hangup extensions execute only once; unable to determine if destination channel up
On 20/11/12 17:14, Richard Mudgett wrote:
This is a question regarding whether there's any way within hangup
extensions to determine whether the caller or callee leg (or both) of
a
bridged call has hung up. The test case I have is running under
Asterisk 1.8.17.0, but the behaviour is observed in 1.8.18.0 (and
also
1.6.2.18).
Within the dialplan, the Dial() application with the F flag, so
that
once the caller hangs up, the dialplan jumps to a new priority which
enables the called party to enter some digits which describe the
outcome
of the call. Also, the g flag is used to attempt to continue
execution of the dialplan if the called party hangs up.
Minimally, the dialplan is covered by the following:
[test]
exten = _1000,1,Set(_CALLER_HUNGUP=false)
exten = _1000,2,Set(_CALLEE_HUNGUP=false)
exten = _1000,3,Dial(SIP/${EXTEN},60,CgF(test^1000^10))
exten = _1000,4,Set(_CALLEE_HUNGUP=true)
exten = _1000,10,Set(_CALLER_HUNGUP=true)
exten = _1000,11,AGI(afterCallWork.agi)
exten = h,1,NoOp(${CALLER_HUNGUP})
exten = h,2,NoOp(${CALLEE_HUNGUP})
exten = h,3,AGI(postCall.agi)
Normally, the hangup extensions execute twice: once when the caller
hangs up, then once more when the called party hangs up, either
during
or after the execution of afterCallWork.agi. This second call is
important so that clean up can be performed.
However, if the two parties hang up simultaneously (or within a
split-second of each other), I often see only one execution of the
hangup extensions. Stranger still, the hangups can occur so close to
each other that execution of the hangup extension occurs without the
either the priority 4 or priority 10 steps being executed (it can be
difficult replicate this, but inserting a Wait(1) call at priority 4
and
another at priority 10 can help here).
In such cases, I see the output from the two NoOps as false and
false.
(This is difficult to replicate because of the precise timing it
requires - it is easy if you insert Wait(1) at priority 4 and 10, but
whether this is valid or not is debatable. I can replicate this
issue
with just the dialplan above on a slowish server).
So I need to be able to query the status of the other channel from
within postCall.agi, because if both parties have hung up, I may only
get one execution of the hangup extensions, and I can go ahead and
perform the cleanup. Is this possible? I've tried calling CHANNEL
STATUS for the destination channel within postCall.agi, but even when
the destination channel is definitely still up, the call returns an
error 511 Command Not Permitted on a dead channel (presumably
because
the current (caller) channel has hung up).
I can't find anything that I can use within the execution of the
hangup
extensions for the caller to determine whether the destination
channel
is still up.
Is it a bug that I only get the one call to the hangup extensions
when
both caller and callee channels hangup so close to each other that
neither the F nor g flags have the desired effect?
No. I don't see this as a bug. Priority 4 and 10 can only execute
while the channel is not hung up. This is normal dialplan execution.
Only the h exten can execute on a hung up channel. Since both
channels are hanging up at the same time, neither priority 4 nor 10
are able to get executed.
The new pre-dial and hangup handler features in Asterisk 11 would
be a solution to your problem. Otherwise, I don't really see a
solution without rethinking your post call processing.
Richard
--
Thanks for the info Richard. I guess we need to be looking at Asterisk 11.
Is there no way to determine whether another channel is up from within
an h exten?
Thanks again
John
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Re: [asterisk-users] leading ghost 0
Then if you did not restart dahdi and asterisk, then the changes to the parameters in chan_dahdi.conf and system.conf were never taken into account. There is no other way than really restarting asterisk and dahdi. Frederic On Wed, 2012-11-21 at 09:08 +0100, gincantalupo wrote: I cannot restart dahdi because the PBX is in production, all I can do is a module reload chan_dahdi.so. -- _ -- 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] leading ghost 0
I am not really sure, restarting asterisk and dahdi can be the most obvious thing to do, but restarting the dahdi kernel module can be useless if you haven't changed the kernel module configuration and reloading the module in asterisk can be enough if you have changed just the chan_dahdi.conf Leandro 2012/11/21 Frederic Van Espen frederic...@gmail.com Then if you did not restart dahdi and asterisk, then the changes to the parameters in chan_dahdi.conf and system.conf were never taken into account. There is no other way than really restarting asterisk and dahdi. Frederic On Wed, 2012-11-21 at 09:08 +0100, gincantalupo wrote: I cannot restart dahdi because the PBX is in production, all I can do is a module reload chan_dahdi.so. -- _ -- 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] watchdog like functions
- Original Message - Switching to SIP is likely your best solution. IAX is buggy. Always has been, and I'll bet always will be. Alright, I'll bite on this one. Can you give any specifics about IAX being buggy, other than throwing out random claims? I understand it doesn't get the industry use and acceptance SIP has seen, but that doesn't automatically discount it's functionality correct? If anything, I've found SIP to be more finicky, mostly due to far end NAT issues or general interop problems. I guess I'm just curious about your IAX experience that would lead you to discount it as 'buggy'. --Tim -- _ -- 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] watchdog like functions
- Original Message - I wish to ask if there is way to keep IAX trunk connection up. I have a small server on Xen VPS but notice that my IAX trunk drops after some time. I understand there is cron job to function as sip watchdog. My asterisk is 11.0.1 You'll want to use 'qualify=yes' for your IAX2 peers which keeps registrations active by sending a 'ping' every 60 seconds (by default). Quite a bit of detail available here: http://www.voip-info.org/wiki/view/Asterisk+iax+qualify --Tim -- _ -- 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] watchdog like functions
asterisk asterisk wrote: I wish to ask if there is way to keep IAX trunk connection up. I have a small server on Xen VPS but notice that my IAX trunk drops after some time. I understand there is cron job to function as sip watchdog. My asterisk is 11.0.1 Can you also define what you mean by IAX trunk drops? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] core show translation - difference in Asterisk Versions
Salman Zafar wrote: Hello All, Hola, I was wondering if somebody could elaborate the change in translation of codecs specifically the amount of time increased in Asterisk 11. For example *_Asterisk 11_* ***alaw **speex * *gsm **15000 **15000 * *ulaw9150 15000* ** *_Asterisk 1.6.x_* ***alaw **speex * *gsm **2 12002 * *ulaw1 12002* I did recalculate the translation for 60 or 1 seconds but nothing changes on Asterisk 11 (VM, Cloud or even physical machine). Is it slin?, adding this overhead or there is something I am overlooking?. I've taken some time to look at what is going on here and let me reassure you - internally stuff is still taking the same amount of time as previously. During a media architecture rework how translations are chosen was changed around a bit to not just take into account computational costs. What you are now seeing above is actually the internal table cost for choosing. I can certainly agree that this is not ideal. The subject has been brought up a few times but nobody has tackled making it reflect computational costs once again. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] leading ghost 0
Alex, I had already tried itreloading chan_dahdi.so module is enough...I saw Asterisk was behaving differently after reload. To tell the truth, setting pridialplan=unknown causes Asterisk to stop reading following channels configuration...it says pridialplan is already unknown so it stops evaluating chan_dahdi.conf file useless to say that all n+1 channels do not work. Maybe it is a bug but with that parameter set in that way I cannot dial. I'm sure Asterisk is dialling the right number: [2012-11-21 09:05:29] VERBOSE[8314] logger.c: [70 0b a1 33 34 39 3x 3x 3x 3x 3x 3x 34] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3497078884' ] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: q931.c:3134 q931_setup: call 32781 on channel 6 enters state 1 (Call Initiated) [2012-11-21 09:05:29] VERBOSE[8314] logger.c: -- Called 6/349xx4 I'm starting to think it is a telco problem... in case I'd change some parameter like pridialplan or similar, shouldn't I just see a leading 0 in the frame like this: [70 0b a1 *30* 33 34 39 3x 3x 3x 3x 3x 3x 34] added by Asterisk/DAHDI?? I've used this page as reference about frame fields: http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm Thank you. Giorgio Incantalupo On 11/20/2012 05:23 PM, Alex Kauffmann wrote: On 11/20/2012 8:03 AM, gincantalupo wrote: Hi Leandro, I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Yes...the technician did it...there is only one cable. Maybe it is the socket circuitry that has something wrong but I do not know ho to check. Asap I'll be on site I'll do more testing. Thank you Giorgio On 11/20/2012 01:13 PM, Leandro Dardini wrote: That is a real mistery! I like a lots these cases when all seems not working despite all being correctly configured, but you know first or later you'll find the answer. From your website, it seems you are selling/renting PBX based on asterisk, so you can be sure nobody has messed with the asterisk or dahdi source code adding a zero... I am sure you have already tried with a brand new server. Have you checked the pridialplan and prilocaldialplan setting? If I was in your shoes, I'll get another server, with a PRI configured as master and hook it at your PBX to really check if the zero is sent. Does the technician try to make phone calls from the same network cable you are using? Leandro 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi Leandro, thanks for your answer. I already have tried those parameters but without any positive result. The telco technician has tried the line with its machine and it worked...remote telco technicians say they get a leading zero... I'm thinking there is something strange in the middle that adds the zero but do not know what it is. Strange is the fact that you can call some numbers with or without the prefix zero... Moreover we had no problem with the previous telco (fastweb). So we can only call PTSN numbersnot mobile phones. Giorgio On 11/20/2012 11:12 AM, Leandro Dardini wrote: 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi all, I have problems dialling out because my new telco (the previous gave no problems) tells me my PBX adds a leading 0 and that's why I cannot dial out (but I can receive calls). I make a small extensions.conf as a test: exten = 666,1,Dial(DAHDI/g1/339xx) but cannot dial out Curious thing is that exten = 666,1,Dial(DAHDI/g1/0233xx) and exten = 666,1,Dial(DAHDI/g1/233xx) call the same number!!! Line in use is a PRI. My Asterisk version is 1.4.26.2 dahdi version: 2.2.0.2 wanpipe-3.4.6 I checked with intense pri debug and see no 0 inside frames How can I really be SURE Asterisk is not adding some leading zero? Thank you. Giorgio. I have never heard of a way to automatically add digits when using PRI, however can you check your chan_dahdi.conf about the following lines: internationalprefix = nationalprefix = localprefix = If presents, try messing with them. If you are using the PRI in Italy, every provider has PRI configured in its own way, some time even the same provider is configuring PRI lines in multiple times, but often the problems are on receiving the calls (like calls with and without the area code, with or without the leading zero, etc. etc.) Leandro -- The prilocaldialplan parameter is for inbound so you
Re: [asterisk-users] leading ghost 0
Any changes inside chan_dahdi requires you to unload module chan_dahdi and load module chan_dahdi, in case you dont wish to.restart asterisk. pridialplan = national or unknown should help you solve the problem, however you need to unload n load dahdi module. Mitul On Nov 21, 2012 10:26 PM, gincantalupo gincantal...@fgasoftware.com wrote: ** Alex, I had already tried itreloading chan_dahdi.so module is enough...I saw Asterisk was behaving differently after reload. To tell the truth, setting pridialplan=unknown causes Asterisk to stop reading following channels configuration...it says pridialplan is already unknown so it stops evaluating chan_dahdi.conf file useless to say that all n+1 channels do not work. Maybe it is a bug but with that parameter set in that way I cannot dial. I'm sure Asterisk is dialling the right number: [2012-11-21 09:05:29] VERBOSE[8314] logger.c: [70 0b a1 33 34 39 3x 3x 3x 3x 3x 3x 34] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3497078884' ] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: q931.c:3134 q931_setup: call 32781 on channel 6 enters state 1 (Call Initiated) [2012-11-21 09:05:29] VERBOSE[8314] logger.c: -- Called 6/349xx4 I'm starting to think it is a telco problem... in case I'd change some parameter like pridialplan or similar, shouldn't I just see a leading 0 in the frame like this: [70 0b a1 *30* 33 34 39 3x 3x 3x 3x 3x 3x 34] added by Asterisk/DAHDI?? I've used this page as reference about frame fields: http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm Thank you. Giorgio Incantalupo On 11/20/2012 05:23 PM, Alex Kauffmann wrote: On 11/20/2012 8:03 AM, gincantalupo wrote: Hi Leandro, I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Yes...the technician did it...there is only one cable. Maybe it is the socket circuitry that has something wrong but I do not know ho to check. Asap I'll be on site I'll do more testing. Thank you Giorgio On 11/20/2012 01:13 PM, Leandro Dardini wrote: That is a real mistery! I like a lots these cases when all seems not working despite all being correctly configured, but you know first or later you'll find the answer. From your website, it seems you are selling/renting PBX based on asterisk, so you can be sure nobody has messed with the asterisk or dahdi source code adding a zero... I am sure you have already tried with a brand new server. Have you checked the pridialplan and prilocaldialplan setting? If I was in your shoes, I'll get another server, with a PRI configured as master and hook it at your PBX to really check if the zero is sent. Does the technician try to make phone calls from the same network cable you are using? Leandro 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com gincantal...@fgasoftware.com Hi Leandro, thanks for your answer. I already have tried those parameters but without any positive result. The telco technician has tried the line with its machine and it worked...remote telco technicians say they get a leading zero... I'm thinking there is something strange in the middle that adds the zero but do not know what it is. Strange is the fact that you can call some numbers with or without the prefix zero... Moreover we had no problem with the previous telco (fastweb). So we can only call PTSN numbersnot mobile phones. Giorgio On 11/20/2012 11:12 AM, Leandro Dardini wrote: 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com gincantal...@fgasoftware.com Hi all, I have problems dialling out because my new telco (the previous gave no problems) tells me my PBX adds a leading 0 and that's why I cannot dial out (but I can receive calls). I make a small extensions.conf as a test: exten = 666,1,Dial(DAHDI/g1/339xx) but cannot dial out Curious thing is that exten = 666,1,Dial(DAHDI/g1/0233xx) and exten = 666,1,Dial(DAHDI/g1/233xx) call the same number!!! Line in use is a PRI. My Asterisk version is 1.4.26.2 dahdi version: 2.2.0.2 wanpipe-3.4.6 I checked with intense pri debug and see no 0 inside frames How can I really be SURE Asterisk is not adding some leading zero? Thank you. Giorgio. I have never heard of a way to automatically add digits when using PRI, however can you check your chan_dahdi.conf about the following lines: internationalprefix
Re: [asterisk-users] leading ghost 0
On 11/21/2012 10:53 AM, gincantalupo wrote: Alex, I had already tried itreloading chan_dahdi.so module is enough...I saw Asterisk was behaving differently after reload. To tell the truth, setting pridialplan=unknown causes Asterisk to stop reading following channels configuration...it says pridialplan is already unknown so it stops evaluating chan_dahdi.conf file useless to say that all n+1 channels do not work. Maybe it is a bug but with that parameter set in that way I cannot dial. I'm sure Asterisk is dialling the right number: [2012-11-21 09:05:29] VERBOSE[8314] logger.c: [70 0b a1 33 34 39 3x 3x 3x 3x 3x 3x 34] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3497078884' ] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: q931.c:3134 q931_setup: call 32781 on channel 6 enters state 1 (Call Initiated) [2012-11-21 09:05:29] VERBOSE[8314] logger.c: -- Called 6/349xx4 I'm starting to think it is a telco problem... in case I'd change some parameter like pridialplan or similar, shouldn't I just see a leading 0 in the frame like this: [70 0b a1 *30* 33 34 39 3x 3x 3x 3x 3x 3x 34] added by Asterisk/DAHDI?? I've used this page as reference about frame fields: http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm Thank you. Giorgio Incantalupo On 11/20/2012 05:23 PM, Alex Kauffmann wrote: On 11/20/2012 8:03 AM, gincantalupo wrote: Hi Leandro, I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Yes...the technician did it...there is only one cable. Maybe it is the socket circuitry that has something wrong but I do not know ho to check. Asap I'll be on site I'll do more testing. Thank you Giorgio On 11/20/2012 01:13 PM, Leandro Dardini wrote: That is a real mistery! I like a lots these cases when all seems not working despite all being correctly configured, but you know first or later you'll find the answer. From your website, it seems you are selling/renting PBX based on asterisk, so you can be sure nobody has messed with the asterisk or dahdi source code adding a zero... I am sure you have already tried with a brand new server. Have you checked the pridialplan and prilocaldialplan setting? If I was in your shoes, I'll get another server, with a PRI configured as master and hook it at your PBX to really check if the zero is sent. Does the technician try to make phone calls from the same network cable you are using? Leandro 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi Leandro, thanks for your answer. I already have tried those parameters but without any positive result. The telco technician has tried the line with its machine and it worked...remote telco technicians say they get a leading zero... I'm thinking there is something strange in the middle that adds the zero but do not know what it is. Strange is the fact that you can call some numbers with or without the prefix zero... Moreover we had no problem with the previous telco (fastweb). So we can only call PTSN numbersnot mobile phones. Giorgio On 11/20/2012 11:12 AM, Leandro Dardini wrote: 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi all, I have problems dialling out because my new telco (the previous gave no problems) tells me my PBX adds a leading 0 and that's why I cannot dial out (but I can receive calls). I make a small extensions.conf as a test: exten = 666,1,Dial(DAHDI/g1/339xx) but cannot dial out Curious thing is that exten = 666,1,Dial(DAHDI/g1/0233xx) and exten = 666,1,Dial(DAHDI/g1/233xx) call the same number!!! Line in use is a PRI. My Asterisk version is 1.4.26.2 dahdi version: 2.2.0.2 wanpipe-3.4.6 I checked with intense pri debug and see no 0 inside frames How can I really be SURE Asterisk is not adding some leading zero? Thank you. Giorgio. I have never heard of a way to automatically add digits when using PRI, however can you check your chan_dahdi.conf about the following lines: internationalprefix = nationalprefix = localprefix = If presents, try messing with them. If you are using the PRI in Italy, every provider has PRI configured in its own way, some time even the same provider is configuring PRI lines in multiple times, but often the problems are on receiving the calls (like calls with and without the area code, with or without the leading zero, etc. etc.) Leandro -- The prilocaldialplan parameter
Re: [asterisk-users] Simple failover configuration
Take a look at this doc from Polycom...it answers your question I think. https://encrypted.google.com/url?sa=trct=jq=polycom%20redundant%20serversource=webcd=1cad=rjaved=0CEUQFjAAurl=http%3A%2F%2Fsupport.polycom.com%2Fglobal%2Fdocuments%2Fsupport%2Ftechnical%2Fproducts%2Fvoice%2FConfiguring_Optional.pdfei=TjGtUMuDD86E0QHPpYCQDAusg=AFQjCNGL4uuttNHorfaTnTGcqxCQAZrwCQsig2=-HbRXBZJR1nqEtT0VmYq1A On Thu, Nov 15, 2012 at 6:59 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: At present I have two hardware identically freepbx/asterisk boxes. The mysql db on one is slaved to the other and all config files are rsync'd once every 24 hours (we have few configuration changes). We use Polycom 321/331/550/650 phones, and I notice that these phones can be configured with two SIP servers. Would the simplest approach to failover be to just configure my primary asterisk server as the first SIP server and my backup as the second? Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Paris - mini-DebConf - VoIP - 24 November
For those using Debian/Ubuntu (and anybody else is welcome of course), there is a mini-DebConf in Paris this weekend: http://fr2012.mini.debconf.org/ There is a presentation at 16:00 about Debian's role in establishing an alternative to Skype, this will look at some of the packages available on the upcoming Debian 7 (wheezy), and strategic ways of deploying them to build a genuinely free and open cloud for real-time communications. There is no registration fee - all welcome -- _ -- 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] Java server side components for asterisk
Hello, I am a java programmer and I will like to integrate Asterisk to my product line. The challenge is that I have been searching for a java server component that I can use to create an Asterisk GUI configurator. it could be open source or commercial . Nweike Onwuyali . excuse my typos brevity. Sent from a mobile device -- _ -- 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
