Re: [asterisk-users] CDR Desgin
I agree with Freddi and would like to add that a field indicating the order of the outgoing legs would be very useful. For billing purposes one could benefit very much if one new the order of the providers that were called in a specific call. Freddi Hansen wrote: To me the obvious answer is to provide a CDR for every call leg so for A calling B via Asterisk there would be two CDRs produced. It's far far easier to disregard the unwanted CDRs than it is to try and generate the missing ones and in some cases it's virtually impossible. If it's weighed up I think people would vote to have accurate CDRs ahead of anything else and if single legs are the best way to do that then it's the way it should be done. In addition with single leg CDRs it will solve the dilemna about acommodating every possible call scenario that I know has caused you a lot of consternation over the last 18 months. Sure it's a change from the current situation so maybe needs to use the standard apporach of a configuration setting to opt in initially before becoming the default in the subsequent major release. OK, Greyman, your logic is solid. If we provide a CDR implementation that just generates the individual call legs, and ties them together via the linkedid (see team/group/newcdr), then both camps should be able to derive the info they need for billing, via hopefully not-overly-complex SQL queries to a backend db. I'll modify my RFC to reflect this line of thinking. Yes, it is a bit of shift. And, yes, the implementation will make this new approach optional, and not default. But, pardon if I make it available via the CEL domain come implementation time. It should take me a week to rehash my document; perhaps longer (I'm in bugfix mode, and this borderline development work); in the meantime, those with decided CDR needs might make their wishes known, if they do not think this approach will work. Speak now, or forever hold your peace; or forever complain... or whatever. If this is particularly distressing to you, perhaps your fears might be slightly assuaged when you read the details... I was part of a team that did design a multiservice billing system about 15 years ago and the solution people seems to agree on here (and me to) looks pretty much the same i.e one call may consist of several calls legs. In addition to the linkedid it would be nice to have an indication in the cdr that tells us that 'this is the lastone on this linked id'. Our experience was that we shouldn't for load reasons work with cdr's in the immidiate multileg format in the DB. So we did collect cdr's in a tmp DB until we got the the record with end marker set. We would then produce our final cdr for the actual service, store it in billing col. and delete it from the multileg col. When a new service is created we only have to make a the new customized cdr, we don't have to touch the generation of the multileg format. Freddi ___ -- 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] Ring/Off-hook in strange state 6 channel X
On Tue, Nov 25, 2008 at 10:26:43PM -0600, [EMAIL PROTECTED] wrote: Greetings List I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all of them give me the error Ring/Off-hook in strange state 6. DAHDI? Zaptel? What version? Whenever the caller hangup, the call continue to execute until it hits the hard coded hangup. I changed chan_dadhi busydetect=no and callprogress=no but problem still persist. I also tried to use different PABX in vain. GSM modem (FUSION100) also produces no useful result Could you include here your chan_dahdi.conf ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Asterisk voicemail and Lotus Notes
Hello, I've seen Domino 7 supports ACL and IMAP. Have you heard of experiences in which Lotus Notes/Domino users could read and manage voicemails recorded by Asterisk ? In other words, is it possible with Domino to dedicate to Asterisk an account with which, using IMAP, Asterisk could drop or remove mails into Lotus Notes users ? Regards ___ -- 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] The sound is played but I did not hear
jhon digital21 wrote: same result I never saw the original message, what version of Asterisk and what country are you in? Does it work for outbout okay? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] SVN
Hello, everyone. Anybody know when that svn will be available again? Regards *Alex Montoanelli* Administração e Gerência de Redes Unetvale Conectividade http://www.unetvale.net +55 48 3263 8700 ___ -- 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] half channel audio after upgrade to 1.4.18
Jerry Geis wrote: / I upgraded from 1.2 to 1.4.18 // // After upgrading I get half channel audio on SOME phones. // // I have Cisco 7960 that works, I have a wireless polycom 8002 phone that // works. // However, my polycom 501's are getting half channel audio on EXTERNAL calls. // Internal calls are OK. // // I have enabled nat=yes on all phones. // // What is something else I can try? Any thoughts on why half channel audio // after the upgrade? // // /'canreinvite=no' can also help with this. PaulH Paul, Many thanks - that did the trick. Jerry ___ -- 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] Hints stopped working suddently
Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- 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] Mobile as FXO
Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) = Asterisk FXO (Nokia 7610) Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone? Regards, Irfan Mali Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk ___ -- 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] Mobile as FXO
A little less whitespace please. If I understand your question correctly, yes you can. On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote: Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone? Regards, Irfan Mali Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Hints stopped working suddently
Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- 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] Channel variable to identify the calling SIP peer
On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady [EMAIL PROTECTED] wrote: Hi folks I'm not sure what I am missing but I cannot find a predefined channel variable to identify the SIP peer/user which has initiated a call and established the channel. The one option is to extract it from the CHANNEL variable, but that is fraught with difficulties. Is there another variable I don't know about or another way to do this? In 1.2 and 1.4 I don't believe there is any other way. Parsing the username from the channel name is what we ended up having to do! Regards, Greyman. ___ -- 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] SVN
On Wed, Nov 26, 2008 at 1:32 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote: Hello, everyone. Anybody know when that svn will be available again? Regards Hey, I can checkout stuff fine from svn.digium.com. Maybe you can provide some more info about how it's not working for you. Probably it's that http://svn.digium.com/ gives 403 error. As i recall, it showed up when some search engine tried to indexing whole SVN ignoring robots.txt, so Digium disabled root page. Now you can access it by adding /view/ to URL. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Dahdi, b410p and looping from 1 port to another
Hello, Is it possible, for testing, to connect an cat5 straight patch cord between 2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk ? I've tried this: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Results: - both ports lights are green - console shows the outgoing call - no call is coming in and nothing happens (no sound, no message on console) Regards ___ -- 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] language and meetme issue
Hello, I have created a dynamic conference into two languages (english and russian). Client calls to confrence number and interactive choose the language. Meetme runs with 'dMi' options. Everything works perfect if one conference room clients have choosed the same language. If clients had choosed different language , there is a problem with user join/leave announcements. For example: First user A choosed language english, after couple seconds joins new conference user B (he choosed language russian), but user A hears new user has joined... announcement in russian language . I want that every user in the same conference number hears announcements in their chosen language (user A hears everything in english, user B in russian) and so on. Is it possible to do that... Thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] Mobile as FXO
Just speaking theoretically, you should be able to do a Zap/SIP bridge just like using a TDM???. How does this show up in the CLI interface (core show channels)? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik Sent: Wednesday, November 26, 2008 8:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Mobile as FXO Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) = Asterisk FXO (Nokia 7610) Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone? Regards, Irfan Mali Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk ___ -- 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] Mobile as FXO
How? Any hint? Regards, Irfan Malik Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, November 26, 2008 7:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mobile as FXO A little less whitespace please. If I understand your question correctly, yes you can. On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote: Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone? Regards, Irfan Mali Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] language and meetme issue
Assuming you have caller id, you can call MeetMe with different parameters. You could also write an AGI to handle the announcements and leave meetme in Silent (No Announce) mode. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys Sent: Wednesday, November 26, 2008 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] language and meetme issue Hello, I have created a dynamic conference into two languages (english and russian). Client calls to confrence number and interactive choose the language. Meetme runs with 'dMi' options. Everything works perfect if one conference room clients have choosed the same language. If clients had choosed different language , there is a problem with user join/leave announcements. For example: First user A choosed language english, after couple seconds joins new conference user B (he choosed language russian), but user A hears new user has joined... announcement in russian language . I want that every user in the same conference number hears announcements in their chosen language (user A hears everything in english, user B in russian) and so on. Is it possible to do that... Thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] Mobile as FXO
What have you tried? On Wed, Nov 26, 2008 at 9:25 AM, Irfan Malik [EMAIL PROTECTED] wrote: How? Any hint? Regards, Irfan Malik Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, November 26, 2008 7:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mobile as FXO A little less whitespace please. If I understand your question correctly, yes you can. On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote: Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone? Regards, Irfan Mali Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Mobile as FXO
What are the lines in your dialplan for using the Mobile line? For example
exten = NXX,1,Dial(Zap/g1/${EXTEN},60)
dials a local (7 digit) number using Zap Group 1, waiting 60 seconds for
connection.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
How? Any hint?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, November 26, 2008 7:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mobile as FXO
A little less whitespace please.
If I understand your question correctly, yes you can.
On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote:
Greetings List,
I have configured chan_mob for Nokia 7610. I can succefully dial from
softphone to mobile and land line numbers,
Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number
When call is established I have to use Nokia 7610 for conversation. Is it
possible to use softphone, dial via mobile phone and have conversation
using
softphone?
Regards,
Irfan Mali
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
___
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To UNSUBSCRIBE or update options visit:
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--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
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Re: [asterisk-users] Mobile as FXO
Here is the output during call, localhost*CLI core show channels Channel Location State Application(Data) Mobile/Nokia-7610-e2 [EMAIL PROTECTED]:1 Ringing AppDial((Outgoing Line)) SIP/2001-09960968[EMAIL PROTECTED]:1 Ring Dial(Mobile/Nokia-7610/0321609 2 active channels 1 active call 3 calls processed localhost*CLI My system hangs when I load zaptel module. Regards, Irfan Malik Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 7:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Mobile as FXO Just speaking theoretically, you should be able to do a Zap/SIP bridge just like using a TDM???. How does this show up in the CLI interface (core show channels)? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik Sent: Wednesday, November 26, 2008 8:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Mobile as FXO Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) = Asterisk FXO (Nokia 7610) Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone? Regards, Irfan Mali Manager MIS TricastMedia Cell +92 321-6099155 PH: +92 42 5785703-8 Ext: 196 Web: www.tcm.com.pk ___ -- 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] Mobile as FXO
These are the lines from my extension.conf
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest) exten = 500,3,Echo
exten = 500,4,Playback(demo-echodone)
exten = 500,5,Hangup
exten = _.,1,Dial(Mobile/Nokia-7610/${EXTEN})
exten = _.,2,Hangup
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, November 26, 2008 7:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mobile as FXO
What have you tried?
On Wed, Nov 26, 2008 at 9:25 AM, Irfan Malik [EMAIL PROTECTED] wrote:
How? Any hint?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, November 26, 2008 7:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mobile as FXO
A little less whitespace please.
If I understand your question correctly, yes you can.
On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED]
wrote:
Greetings List,
I have configured chan_mob for Nokia 7610. I can succefully dial from
softphone to mobile and land line numbers,
Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number
When call is established I have to use Nokia 7610 for conversation. Is it
possible to use softphone, dial via mobile phone and have conversation
using
softphone?
Regards,
Irfan Mali
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
___
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--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
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Re: [asterisk-users] The sound is played but I did not hear
Doug Lytle wrote: jhon digital21 wrote: same result country are you in? Does it work for outbout okay? That should have read 'outbound', that's what happens when you reply when you're late for work. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] MS Exchange IMAP Voicemail
Hi Andrew and all those following this thread; I have gotten it working like it was meant to work see my original post quoted below. I have also included the direct link to my post... My Original Post: http://lists.digium.com/pipermail/asterisk-users/2008-November/222339.ht ml Quote: BTW... I have only tested this on Exchange 2003, I have not yet had the chance to check it out on Exchange 2007, but I'm guessing that it works... I will update when I know... Thanks, Jeff Phelps IT Support Specialist Hi Noah, Yes, there is a way with Exchange 2003 to use a master user. After doing lots of IMAP hacking and testing on Exchange 2003, I found that there IS A WAY!!! I am using Asterisk 1.6.1-Beta2, but this should also work in 1.4.x as it is Exchange specific, not Asterisk specific. I'm sure this is the long awaited for secret that many IT Professionals have been looking for and here is how it works... In your voicemail.conf: ext_num = vm_pass,user_name,user_email,user_pager_email|imapuser=domain.com\admin_ user_name\mailbox_name|imappassword=apmin_user_password The admin username is just the username, and the mailbox name is just the prefix (before the @ symbol) of the e-mail address. Example: 1688 = 1234,1688, http://lists.digium.com/mailman/listinfo/asterisk-users [EMAIL PROTECTED],,tz=central|imapuser=domain.com\vmadmin\user|imappasswor d=Asterisk123 It works for me, let me know if it works for the rest of you!!! Thanks, Jeff Phelps IT Support Specialist ___ -- 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] Mobile as FXO
I would try this:
exten = _.,1,Dial(Mobile/Nokia-7610/${EXTEN},60,KkTt) ; dials using mobile
nokia
7610
This should make the call Bridgeable/Transferrable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
These are lines from my extensions.conf
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest)
exten = 500,3,Echo
exten = 500,4,Playback(demo-echodone)
exten = 500,5,Hangup
exten = _.,1,Dial(Mobile/Nokia-7610/${EXTEN}) ; dials using mobile nokia
7610
exten = _.,2,Hangup
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
Sent: Wednesday, November 26, 2008 7:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
What are the lines in your dialplan for using the Mobile line? For example
exten = NXX,1,Dial(Zap/g1/${EXTEN},60)
dials a local (7 digit) number using Zap Group 1, waiting 60 seconds for
connection.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
How? Any hint?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, November 26, 2008 7:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mobile as FXO
A little less whitespace please.
If I understand your question correctly, yes you can.
On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote:
Greetings List,
I have configured chan_mob for Nokia 7610. I can succefully dial from
softphone to mobile and land line numbers,
Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number
When call is established I have to use Nokia 7610 for conversation. Is it
possible to use softphone, dial via mobile phone and have conversation
using
softphone?
Regards,
Irfan Mali
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
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Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another
Olivier wrote: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami What is your configuration in chan_dahdi.conf? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] The sound is played but I did not hear
Asterisk version : 1.4 country : France outbound : not tested 2008/11/26 Doug Lytle [EMAIL PROTECTED] Doug Lytle wrote: jhon digital21 wrote: same result country are you in? Does it work for outbout okay? That should have read 'outbound', that's what happens when you reply when you're late for work. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] SVN
On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote: Hello, everyone. Anybody know when that svn will be available again? Regards Hey, I can checkout stuff fine from svn.digium.com. Maybe you can provide some more info about how it's not working for you. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] language and meetme issue
2008/11/26 Danny Nicholas [EMAIL PROTECTED] Assuming you have caller id, you can call MeetMe with different parameters. You could also write an AGI to handle the announcements and leave meetme in Silent (No Announce) mode. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Giedrius Augys *Sent:* Wednesday, November 26, 2008 8:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] language and meetme issue Hello, I have created a dynamic conference into two languages (english and russian). Client calls to confrence number and interactive choose the language. Meetme runs with 'dMi' options. Everything works perfect if one conference room clients have choosed the same language. If clients had choosed different language , there is a problem with user join/leave announcements. For example: First user A choosed language english, after couple seconds joins new conference user B (he choosed language russian), but user A hears new user has joined... announcement in russian language . I want that every user in the same conference number hears announcements in their chosen language (user A hears everything in english, user B in russian) and so on. Is it possible to do that... Thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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 Hi, This conference number would be public. So I don't know caller numbers... And I don't think so, that AGI script can help. I think, I need to modify app_meetme application. -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] Channel variable to identify the calling SIP peer
Hi folks I'm not sure what I am missing but I cannot find a predefined channel variable to identify the SIP peer/user which has initiated a call and established the channel. The one option is to extract it from the CHANNEL variable, but that is fraught with difficulties. Is there another variable I don't know about or another way to do this? Thanks in advance! Richard -- Richard Brady T: +44 (0)7771 623 348 E: [EMAIL PROTECTED] ___ -- 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] Mobile as FXO
These are lines from my extensions.conf
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest)
exten = 500,3,Echo
exten = 500,4,Playback(demo-echodone)
exten = 500,5,Hangup
exten = _.,1,Dial(Mobile/Nokia-7610/${EXTEN}) ; dials using mobile nokia
7610
exten = _.,2,Hangup
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
Sent: Wednesday, November 26, 2008 7:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
What are the lines in your dialplan for using the Mobile line? For example
exten = NXX,1,Dial(Zap/g1/${EXTEN},60)
dials a local (7 digit) number using Zap Group 1, waiting 60 seconds for
connection.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
How? Any hint?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, November 26, 2008 7:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mobile as FXO
A little less whitespace please.
If I understand your question correctly, yes you can.
On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote:
Greetings List,
I have configured chan_mob for Nokia 7610. I can succefully dial from
softphone to mobile and land line numbers,
Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number
When call is established I have to use Nokia 7610 for conversation. Is it
possible to use softphone, dial via mobile phone and have conversation
using
softphone?
Regards,
Irfan Mali
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
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--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another
Hi, From your answer, shall I understand it is possible to loop for one port back to another ? Anyway, chan_dahdi.conf : [channels] language=fr context=isdntrunk switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=bri_cpe_ptp channel=1-2 channel=4-5 ___ -- 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] language and meetme issue
Ok. You will need to modify meetme.c to allow a prompt for language as well as name. Based on the prompt, you will provide the chosen language to the asterisk say prompts in the routine. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys Sent: Wednesday, November 26, 2008 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] language and meetme issue 2008/11/26 Danny Nicholas [EMAIL PROTECTED] Assuming you have caller id, you can call MeetMe with different parameters. You could also write an AGI to handle the announcements and leave meetme in Silent (No Announce) mode. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys Sent: Wednesday, November 26, 2008 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] language and meetme issue Hello, I have created a dynamic conference into two languages (english and russian). Client calls to confrence number and interactive choose the language. Meetme runs with 'dMi' options. Everything works perfect if one conference room clients have choosed the same language. If clients had choosed different language , there is a problem with user join/leave announcements. For example: First user A choosed language english, after couple seconds joins new conference user B (he choosed language russian), but user A hears new user has joined... announcement in russian language . I want that every user in the same conference number hears announcements in their chosen language (user A hears everything in english, user B in russian) and so on. Is it possible to do that... Thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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 Hi, This conference number would be public. So I don't know caller numbers... And I don't think so, that AGI script can help. I think, I need to modify app_meetme application. -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] SVN
I was trying a 'svn ls http://svn.digium.com/svn/', and was receiving a 403 - Forbiden. But a rising level could access the content. Thank you and hugs Regards *Alex Montoanelli* On Wed, Nov 26, 2008 at 12:17 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Wed, Nov 26, 2008 at 1:32 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote: Hello, everyone. Anybody know when that svn will be available again? Regards Hey, I can checkout stuff fine from svn.digium.com. Maybe you can provide some more info about how it's not working for you. Probably it's that http://svn.digium.com/ gives 403 error. As i recall, it showed up when some search engine tried to indexing whole SVN ignoring robots.txt, so Digium disabled root page. Now you can access it by adding /view/ to URL. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Hints stopped working suddently
For me, the Polycom loses its subscription when asterisk is restarted. However, as long as the phone is restarted after asterisk, everything works fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than yours.) ___ -- 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] 1.4.x Strange Vocemail delay
Hi there,
I've got the following code (for remote enquiry of the answering machine) in
my dialplan:
[mailbox]
exten = m,1,Set(TIMEOUT(digit)=4)
exten = m,2,Set(TIMEOUT(response)=0)
exten = m,3,Set(LANGUAGE()=de)
exten = m,4,Read(Pin,unavail,4)
exten = m,5,capicommand(echosquelch|no)
exten = m,6,Gotoif($[${Pin} = ${MBPIN}]?7:9)
exten = m,7,VoicemailMain([EMAIL PROTECTED]|s)
exten = m,8,Hangup
exten = m,9,Voicemail([EMAIL PROTECTED]|s)
exten = m,10,Hangup
exten = t,11,Voicemail([EMAIL PROTECTED]|s)
exten = t,12,Hangup
While this used to work fine in Asterisk 1.2.x from debian stable (etch) it
does not work as expected with Asterisk 1.4 from Debian testing (lenny)
anymore.
There is some strange delay between the the End of the Read command and the
point where 'beep' is going to get played.
Whats going on in this case with 1.4?
Sven
--
The main thing to note is that when you choose open source you don't
get a Windows operating system.
(from http://www.dell.com/ubuntu)
/me is [EMAIL PROTECTED], http://sven.gegg.us/ on the Web
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Re: [asterisk-users] Hints stopped working suddently
The phone should renew itself to asterisk periodically even after a reboot. My setup renews the connection every 2 minutes (non-critical, small shop). _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Wednesday, November 26, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently For me, the Polycom loses its subscription when asterisk is restarted. However, as long as the phone is restarted after asterisk, everything works fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than yours.) ___ -- 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] language and meetme issue
2008/11/26 Danny Nicholas [EMAIL PROTECTED] Ok. You will need to modify meetme.c to allow a prompt for language as well as name. Based on the prompt, you will provide the chosen language to the asterisk say prompts in the routine. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Giedrius Augys *Sent:* Wednesday, November 26, 2008 9:04 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] language and meetme issue 2008/11/26 Danny Nicholas [EMAIL PROTECTED] Assuming you have caller id, you can call MeetMe with different parameters. You could also write an AGI to handle the announcements and leave meetme in Silent (No Announce) mode. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Giedrius Augys *Sent:* Wednesday, November 26, 2008 8:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] language and meetme issue Hello, I have created a dynamic conference into two languages (english and russian). Client calls to confrence number and interactive choose the language. Meetme runs with 'dMi' options. Everything works perfect if one conference room clients have choosed the same language. If clients had choosed different language , there is a problem with user join/leave announcements. For example: First user A choosed language english, after couple seconds joins new conference user B (he choosed language russian), but user A hears new user has joined... announcement in russian language . I want that every user in the same conference number hears announcements in their chosen language (user A hears everything in english, user B in russian) and so on. Is it possible to do that... Thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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 Hi, This conference number would be public. So I don't know caller numbers... And I don't think so, that AGI script can help. I think, I need to modify app_meetme application. -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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 It would be hard but possible :) -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] Customized CDR Records
You can compile this code into Asterisk 1.4 to give you the ability to write
custom data for up to 20 fields. The field names in the code must match the
field names in the cdr db table. ENJOY
Dave
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2003-2005, Digium, Inc.
*
* Brian K. West [EMAIL PROTECTED]
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief ODBC CDR Backend
*
* \author Brian K. West [EMAIL PROTECTED]
*
* See also:
* \arg http://www.unixodbc.org
* \arg \ref Config_cdr
* \ingroup cdr_drivers
*/
/*** MODULEINFO
dependunixodbc/depend
dependltdl/depend
***/
#include asterisk.h
ASTERISK_FILE_VERSION(__FILE__, $Revision: 69702 $)
#include sys/types.h
#include stdio.h
#include string.h
#include stdlib.h
#include unistd.h
#include time.h
#ifndef __CYGWIN__
#include sql.h
#include sqlext.h
#include sqltypes.h
#else
#include windows.h
#include w32api/sql.h
#include w32api/sqlext.h
#include w32api/sqltypes.h
#endif
#include asterisk/config.h
#include asterisk/options.h
#include asterisk/channel.h
#include asterisk/cdr.h
#include asterisk/module.h
#include asterisk/logger.h
#define DATE_FORMAT %Y-%m-%d %T
static char *name = ODBC;
static char *config = cdr_odbc.conf;
static char *dsn = NULL, *username = NULL, *password = NULL, *table = NULL;
static int loguniqueid = 0;
static int usegmtime = 0;
static int dispositionstring = 0;
static int connected = 0;
AST_MUTEX_DEFINE_STATIC(odbc_lock);
static int odbc_do_query(void);
static int odbc_init(void);
static SQLHENV ODBC_env = SQL_NULL_HANDLE; /* global ODBC Environment */
static SQLHDBC ODBC_con; /* global ODBC Connection
Handle */
static SQLHSTMT ODBC_stmt; /* global ODBC Statement Handle
*/
static void odbc_disconnect(void)
{
SQLDisconnect(ODBC_con);
SQLFreeHandle(SQL_HANDLE_DBC, ODBC_con);
SQLFreeHandle(SQL_HANDLE_ENV, ODBC_env);
connected = 0;
}
static char* getthevar(struct ast_cdr *cdr, char *name, char *buf, int size)
{
struct ast_channel dummy;
char format[256] = \0;
memset(buf, 0, size);
/* Quite possibly the first use of a static struct ast_channel, we need
it so the var funcs will work */
memset(dummy, 0, sizeof(dummy));
dummy.cdr = cdr;
snprintf(format, sizeof(format) - 1, ${CDR(%s)}, name);
pbx_substitute_variables_helper(dummy, format, buf, size);
ast_log(LOG_NOTICE, ODBC: Sub value - %s=%s\n, format, buf);
return buf;
}
static int odbc_log(struct ast_cdr *cdr)
{
int ODBC_res;
char sqlcmd[2048] = , timestr[128];
/* This is a really ugly way of doing this, but had to do it quick!! */
char buf[256] = \0;
char buf1[256] = \0;
char buf2[256] = \0;
char buf3[256] = \0;
char buf4[256] = \0;
char buf5[256] = \0;
char buf6[256] = \0;
char buf7[256] = \0;
char buf8[256] = \0;
char buf9[256] = \0;
char buf10[256] = \0;
char buf11[256] = \0;
char buf12[256] = \0;
char buf13[256] = \0;
char buf14[256] = \0;
char buf15[256] = \0;
char buf16[256] = \0;
char buf17[256] = \0;
char buf18[256] = \0;
char buf19[256] = \0;
int res = 0;
struct tm tm;
ast_log(LOG_NOTICE, ODBC: Starting\n);
if (usegmtime)
gmtime_r(cdr-start.tv_sec,tm);
else
ast_localtime(cdr-start.tv_sec, tm, NULL);
ast_mutex_lock(odbc_lock);
strftime(timestr, sizeof(timestr), DATE_FORMAT, tm);
memset(sqlcmd,0,2048);
if (loguniqueid) {
snprintf(sqlcmd,sizeof(sqlcmd),INSERT INTO %s
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,
lastdata,duration,billsec,disposition,amaflags,accountcode,
Field1,Field2,Field3,Field4,Field5,Field6,Field7,Field8,
Field9,Field10,Field11,Field12,Field13,Field14,
Field15,Field16,Field17,Field18,Field19,Field20,uniqueid,userfield)
VALUES
(?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?),
table);
} else {
snprintf(sqlcmd,sizeof(sqlcmd),INSERT INTO %s
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,
duration,billsec,disposition,amaflags,accountcode,
Field1,Field2,Field3,Field4,Field5,Field6,Field7,Field8,
Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another
On Wed, Nov 26, 2008 at 04:12:58PM +0100, Olivier wrote: Hi, From your answer, shall I understand it is possible to loop for one port back to another ? /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Hmm... which of those two should provide timing? I suppose you should use something of the sort of: span=1,1,0,ccs,ami span=2,0,0,ccs,ami Anyway, chan_dahdi.conf : [channels] language=fr context=isdntrunk switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=bri_cpe_ptp channel=1-2 channel=4-5 s/bri_cpe_ptp/bri_cpe/ In addition to that, a CPE needs to talk to a Network on the other side. Thus you should have something of the sort of: signalling=bri_net channel=1-2 signalling=bri_cpe channel=4-5 (As for the question of wiring: I have no idea. Refer to the documentation or to the answers of others in this thread) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] spandsp not recognized by menuselect on Lenny
I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny. I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 : apt-get install libspandsp1 cd /usr/src/asterisk-1.6.0.1 ./configure make clean make menuselect In menuselect/application menu, I can see that app_fax is greyed out and cannot be selected. 1. My understanding is that both spandsp and libtiff4 are not daemons and do not need any configuration, right ? 2. How can I check spandsp is correctly installed and should be made available by menuselect ? Note: with apt-get install libspandsp1 on Lenny, spandsp-0.0.5pre4-1 and libtiff4 are installed and configured. Cheers ___ -- 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] Hints stopped working suddently
Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- 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] Hints stopped working suddently
Good theory, but I had already tried that (and my phone re-subscribes every 60 seconds anyways) so that's not it. Regards, Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Wednesday, November 26, 2008 10:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently For me, the Polycom loses its subscription when asterisk is restarted. However, as long as the phone is restarted after asterisk, everything works fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than yours.) ___ -- 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] Hints stopped working suddently
Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- 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] sip MWI Messages-Waiting: always reports no messages
Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like [EMAIL PROTECTED] in sip.conf, but that doesn't help any. I also tried the same with the mailbox containing messages under [default], but still no luck. I see messages like this if I do sip set debug ip 10.5.7.130 - Reliably Transmitting (no NAT) to 10.5.7.130:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.7.21:5060;branch=z9hG4bK44627853;rport From: asterisk sip:[EMAIL PROTECTED];tag=as7d9b65d4 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 84 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- - from sip.conf: -- [29] insecure=port,invite context=ccme type=friend host=r2.home.misty.com qualify=yes dtmfmode=rfc2833 canreinvite=no nat=no mailbox=29 fromuser=777 vmexten=777 username=29 from voicemail.conf: --- [home] ; testing 29 = 1234,Joe Test,[EMAIL PROTECTED] --- [EMAIL PROTECTED] asterisk]# ls /var/spool/asterisk/voicemail/home/29/INBOX msg.gsm msg.WAV msg0001.wav msg0002.txt msg0003.gsm msg0003.WAV msg0004.wav msg.txt msg0001.gsm msg0001.WAV msg0002.wav msg0003.txt msg0004.gsm msg0004.WAV msg.wav msg0001.txt msg0002.gsm msg0002.WAV msg0003.wav msg0004.txt -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.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
Re: [asterisk-users] Hints stopped working suddently
Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 11:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- 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] Dahdi, b410p and looping from 1 port to another
2008/11/26 Tzafrir Cohen [EMAIL PROTECTED] On Wed, Nov 26, 2008 at 04:12:58PM +0100, Olivier wrote: Hi, From your answer, shall I understand it is possible to loop for one port back to another ? /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Hmm... which of those two should provide timing? I suppose you should use something of the sort of: span=1,1,0,ccs,ami span=2,0,0,ccs,ami done : changed to span=1,1,0,ccs,ami span=2,0,0,ccs,ami (if my memory is ok, port 1 is set to NT mode, port 2 is to TE. I'm still wondering how to teach to port 2 to use timing from port 1 (I think span=2,0 means that but 'm not certain) and above all, how to teach port 1 use a timing source elsewhere and use it) Anyway, chan_dahdi.conf : [channels] language=fr context=isdntrunk switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=bri_cpe_ptp channel=1-2 channel=4-5 s/bri_cpe_ptp/bri_cpe/ Oops ! In addition to that, a CPE needs to talk to a Network on the other side. Thus you should have something of the sort of: signalling=bri_net channel=1-2 signalling=bri_cpe channel=4-5 (As for the question of wiring: I have no idea. Refer to the documentation or to the answers of others in this thread) b410 manual says pins are affected this way : 3 Tx+ (TE) Rx+ (NT) 4 Rx+(TE)Tx+ (NT) 5 Rx-(TE)Tx- (NT) 6 Tx- (TE) Rx- (NT) I don't know if this means a straight cable is ok or not. If someone could shed some light ... -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Customized CDR Records
On Wednesday 26 November 2008 09:48:40 David Budny wrote: You can compile this code into Asterisk 1.4 to give you the ability to write custom data for up to 20 fields. The field names in the code must match the field names in the cdr db table. ENJOY Or you could just use the cdr_adaptive_odbc backport for 1.4: http://svncommunity.digium.com/view/tilghman/branches/1.4/ -- Tilghman ___ -- 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] Dahdi, b410p and looping from 1 port to another
Olivier wrote: (As for the question of wiring: I have no idea. Refer to the documentation or to the answers of others in this thread) b410 manual says pins are affected this way : 3 Tx+ (TE) Rx+ (NT) 4 Rx+(TE)Tx+ (NT) 5 Rx-(TE)Tx- (NT) 6 Tx- (TE) Rx- (NT) I don't know if this means a straight cable is ok or not. If someone could shed some light ... It should be fairly obvious from that documentation. The entire purpose of the jumpers is to switch between NT wiring and TE wiring, and on page 18 of the manual (currently on www.digium.com) it specifically says This eliminates the need to use a crossover cable. Consider the light to be shed :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] The sound is played but I did not hear
jhon digital21 wrote:
Asterisk version : 1.4
country : France
outbound : not tested
Someone else may need to chime in here, I'm in the US.
But, when I was doing analog (PRI all the way around now), I used to use
ztmonitor to measure in inbound/outbound volume. You may want to try
that tool as well.
ztmonitor {chan.number} -v
For example, monitoring channel 1 would be
ztmonitor 1 -v
It will give you a graphical represenatation of the inbound/outbound
active call.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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[asterisk-users] Problems with Rhino Channelbank...
I am having an issue with a Rhino channelbank connected to a Digium TE411P card. The server has 3 E1 R2 links and the fourth port is used to connect a Rhino FXO channelbank with 12 lines. The first four ports on the rhino are GSM adapters. From time to time I can see the channels answering when there is no incoming call. It happens with all the adapters but not on regular phone lines connected to the same unit. What could be the cause? A second issue was that some people were reporting eco on the local side when making calls to mobile phones using the GSM adapters. Since the Te411P has hardware echo cancellation I wonder how this is possible? We are using Asterisk 1.4.22 with Zaptel 1.4.12.1. The lines on the channelbank are configured as fxsks (I have tried fxsls with the same results). -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] spandsp not recognized by menuselect on Lenny
On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote: I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny. I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 : apt-get install libspandsp1 cd /usr/src/asterisk-1.6.0.1 ./configure make clean make menuselect In menuselect/application menu, I can see that app_fax is greyed out and cannot be selected. 1. My understanding is that both spandsp and libtiff4 are not daemons and do not need any configuration, right ? 2. How can I check spandsp is correctly installed and should be made available by menuselect ? Note: with apt-get install libspandsp1 on Lenny, spandsp-0.0.5pre4-1 and libtiff4 are installed and configured. aptitude install libspandsp-dev -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] pick up IAX2 calls
The problem is that IAX2 does not seem to support call pickup. Bruno Castelo Branco wrote: hi I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 for all IAX extensions in iax.conf. Didn't works for while. thanks Tim Panton wrote: I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I think that if you have a Dial() that rings several extens, then any of the technologies involved can pickup with *8 So if you have Dial(IAX/fredSIP/billzap/mark) then someone in the same group as fred can pickup with IAX and someone in the same group as bill can pickup with SIP etc. So it's an asterisk thing, not an IAX thing per-se. Tim. P.S. (you could try putting in a dummy 'fred' entry into Dial and iax.conf.) T. On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote: hi thanks Luis , but doesn't work. For SIP extensions works well *8, but for IAX a tried *8 and ** + iax extension and didn't works Luis Morales wrote: Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- 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 -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- 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] spandsp not recognized by menuselect on Lenny
Tzafrir Cohen schrieb: On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote: I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny. I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 : apt-get install libspandsp1 cd /usr/src/asterisk-1.6.0.1 ./configure make clean make menuselect In menuselect/application menu, I can see that app_fax is greyed out and cannot be selected. 1. My understanding is that both spandsp and libtiff4 are not daemons and do not need any configuration, right ? 2. How can I check spandsp is correctly installed and should be made available by menuselect ? Note: with apt-get install libspandsp1 on Lenny, spandsp-0.0.5pre4-1 and libtiff4 are installed and configured. aptitude install libspandsp-dev Yup. libspandsp is a build-dependency (in contrast to a normal runtime dependency). Thus you need the -dev package. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] spandsp not recognized by menuselect on Lenny
On Wed, Nov 26, 2008 at 07:16:05PM +0100, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote: I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny. I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 : apt-get install libspandsp1 cd /usr/src/asterisk-1.6.0.1 ./configure make clean make menuselect In menuselect/application menu, I can see that app_fax is greyed out and cannot be selected. 1. My understanding is that both spandsp and libtiff4 are not daemons and do not need any configuration, right ? 2. How can I check spandsp is correctly installed and should be made available by menuselect ? Note: with apt-get install libspandsp1 on Lenny, spandsp-0.0.5pre4-1 and libtiff4 are installed and configured. aptitude install libspandsp-dev Yup. libspandsp is a build-dependency (in contrast to a normal runtime dependency). Thus you need the -dev package. Up-to-date list of build dependencies: http://svn.debian.org/viewsvn/pkg-voip/asterisk/branches/experimental/debian/control?rev=6480view=markup dahdi-linux, dahdi-tools (required), libpri 1.4.4 and libss7 are not in Lenny. Libss7 is currently in the new queue. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Dahdi, b410p and looping from 1 port to another
2008/11/26 Kevin P. Fleming [EMAIL PROTECTED] Olivier wrote: (As for the question of wiring: I have no idea. Refer to the documentation or to the answers of others in this thread) b410 manual says pins are affected this way : 3 Tx+ (TE) Rx+ (NT) 4 Rx+(TE)Tx+ (NT) 5 Rx-(TE)Tx- (NT) 6 Tx- (TE) Rx- (NT) I don't know if this means a straight cable is ok or not. If someone could shed some light ... It should be fairly obvious from that documentation. The entire purpose of the jumpers is to switch between NT wiring and TE wiring, and on page 18 of the manual (currently on www.digium.com) it specifically says This eliminates the need to use a crossover cable. Consider the light to be shed :-) Yes but both jumpers were set according this doc. But as we're talking about jumpers, after reading this same doc (page 19) I then had a doubt about whether or not I should turn on100ohm termination to NT port. Doc says it should be turned on in those instances where a BRI is daisy-chained and terminated on the B410P in NT mode. So after reading this again, I would I should have turned it on. Do you agree ? B410P card1 - port1 - TE mode -- B410P card2 - port1 - NT mode -- B410P card2 - port2 - NT mode -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] spandsp not recognized by menuselect on Lenny
2008/11/26 Philipp Kempgen [EMAIL PROTECTED] Yup. libspandsp is a build-dependency (in contrast to a normal runtime dependency). Thus you need the -dev package. Philipp Kempgen I was not aware of such build-dependency packages. Does that mean I could remove build-dependency packages once building is done ? ___ -- 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] spandsp not recognized by menuselect on Lenny [SOLVED]
2008/11/26 Tzafrir Cohen [EMAIL PROTECTED] aptitude install libspandsp-dev It did it ! Thanks, very much. ___ -- 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] spandsp not recognized by menuselect on Lenny
Tzafrir Cohen schrieb: On Wed, Nov 26, 2008 at 07:16:05PM +0100, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote: I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny. I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 : apt-get install libspandsp1 cd /usr/src/asterisk-1.6.0.1 ./configure make clean make menuselect In menuselect/application menu, I can see that app_fax is greyed out and cannot be selected. 1. My understanding is that both spandsp and libtiff4 are not daemons and do not need any configuration, right ? 2. How can I check spandsp is correctly installed and should be made available by menuselect ? Note: with apt-get install libspandsp1 on Lenny, spandsp-0.0.5pre4-1 and libtiff4 are installed and configured. aptitude install libspandsp-dev Yup. libspandsp is a build-dependency (in contrast to a normal runtime dependency). Thus you need the -dev package. Up-to-date list of build dependencies: http://svn.debian.org/viewsvn/pkg-voip/asterisk/branches/experimental/debian/control?rev=6480view=markup BTW: That list doesn't include libspandsp* :-) Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] spandsp not recognized by menuselect on Lenny
Olivier schrieb: 2008/11/26 Philipp Kempgen [EMAIL PROTECTED] Yup. libspandsp is a build-dependency (in contrast to a normal runtime dependency). Thus you need the -dev package. I was not aware of such build-dependency packages. Does that mean I could remove build-dependency packages once building is done ? I think you could uninstall most (if not all) of the *-dev packages after building Asterisk. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] resample not recognized by menuselect on Lenny
BTW (sorry for hijacking the thread): What package satisfies the dependency on resample on Debian Lenny? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] sip MWI Messages-Waiting: always reports no messages
In sip.conf do you have [EMAIL PROTECTED] Lincoln From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Mark G. Thomas [EMAIL PROTECTED] Sent: Wednesday, November 26, 2008 11:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip MWI Messages-Waiting: always reports no messages Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like [EMAIL PROTECTED] in sip.conf, but that doesn't help any. I also tried the same with the mailbox containing messages under [default], but still no luck. I see messages like this if I do sip set debug ip 10.5.7.130 - Reliably Transmitting (no NAT) to 10.5.7.130:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.7.21:5060;branch=z9hG4bK44627853;rport From: asterisk sip:[EMAIL PROTECTED];tag=as7d9b65d4 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 84 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- - from sip.conf: -- [29] insecure=port,invite context=ccme type=friend host=r2.home.misty.com qualify=yes dtmfmode=rfc2833 canreinvite=no nat=no mailbox=29 fromuser=777 vmexten=777 username=29 from voicemail.conf: --- [home] ; testing 29 = 1234,Joe Test,[EMAIL PROTECTED] --- [EMAIL PROTECTED] asterisk]# ls /var/spool/asterisk/voicemail/home/29/INBOX msg.gsm msg.WAV msg0001.wav msg0002.txt msg0003.gsm msg0003.WAV msg0004.wav msg.txt msg0001.gsm msg0001.WAV msg0002.wav msg0003.txt msg0004.gsm msg0004.WAV msg.wav msg0001.txt msg0002.gsm msg0002.WAV msg0003.wav msg0004.txt -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.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
Re: [asterisk-users] Asterisk daemon dies about once per day
OK. I know it's been a few weeks since my original post. Things have been busy ;-) Based on help from the trixbox forums and the asterisk-users mailing list, I have located 30 asterisk core dump files in /tmp. These date from 10/30/08 to 10/24/08. Today is 10/26/08. So this does agree with the intermittent nature of the problem. Many days there are no dumps. Other days there are 5, 7, 6, 1, 4, or 2 dumps. I have used the viewcore tool as indicated on http://www.voip-info.org/wiki-Asterisk+debugging on one of the most recent dump files and posted the output here: http://kgotsi.com/static.php?page=static-asterisk-core-dumps I really don't know what all of the output it produced means, so I'm relying on others with more expertise here to take a look and tell me what insight it may provide. FYI, I have not recompiled asterisk yet. I'm a little nervous about downtime that could be caused by the process not going smoothly, but definitely willing to do so if that's what is needed to fix this problem. Also today I changed a setting in FreePBX, which may have disabled the fax functionality (as previously mentioned as least one other trixbox user who had a problem similar to mine got it fixed by disabling faxing). The setting I changed was in the General Settings, under Fax Machine, changing the Extension of fax machine for receiving faxes from system to disabled. I do not know whether this effectively disables faxing, but it looks like it may. Also, I have not yet used the strace tool. I did install it from the CentOS repositories, and tried to use it, but I didn't have the syntax right. I may want to see if someone can assist me with the correct syntax to run it against my dump files (I believe this is what I'm supposed to do. Please let me know if I'm wrong here). I have also looked at one of the asterisk full log files up to the point that the daemon died. It appears to me that this could also point to the fax functionality causing the issue. I have posted the 200 lines prior to the daemon dying at: http://kgotsi.com/static.php?page=static-asterisk-full-log Again, any insight that someone could provide by examining this would be greatly appreciated. So I guess this is about all of the information I have to post for now. Thanks in advance for any assistance. Sincerely, - Doug Mortensen Network Consultant Impala Networks Original Message -- Message: 8 Date: Mon, 10 Nov 2008 12:52:22 -0700 From: Steve Murphy [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk daemon dies about once per day To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Mon, 2008-11-10 at 10:50 -0700, Steve Murphy wrote: On Mon, 2008-11-10 at 10:10 -0700, Douglas Mortensen wrote: I have an asterisk system where the asterisk daemon dies typically at least once per day. It is running in the wrapper safe_asterisk, which automatically starts the daemon back up. But we find this unacceptable because when the daemon dies, we usually have active calls drop, and sometimes we have to run asterisk -r -x module reload after the daemon starts back up before everything is working well again. Any help or insight would be greatly appreciated. Douglas-- Are you getting core files in /tmp? Getting a stack trace from them could be very informative! If not, or there is no debug info in your asterisk, then I encourage you to recompile asterisk so that DONT_OPTIMIZE is turned off; and so your uhhh, I mean turned ON... sorry safe_asterisk script uses the g option to start asterisk, so it dumps core on a crash. murf Here's an overview of our system. Software Distro: Trixbox CE 2.6.1.1 (CentOS 5) Linux Kernel: 2.6.18-53.1.4.el5 Asterisk version: 1.4.21.2-2 (trixbox RPM) asterisk-addons: 1.4.6 (trixbox RPM) zaptel 1.4.11-1 (trixbox RPM) zaptel-modules 1.4.11-1.2.6.18_53.1.4.el5 (trixbox RPM) Hardware Rhino Ceros III (2U short-depth server) - Intel Desktop Board DG33FB - Intel Pentium D 2.2Ghz (E2200) - 1GB RAM - 80GB SATA HostRAID Mirror (RAID1) - Rhino R1T1-EC Single T1 card (as PRI, using 4 channels + D) - Rhino RCB8FXX/1 w 1 FXO Module (2 FXO ports total) Zaptel The cards we are using are mentioned above. Other than that, if it helps, here's what we're doing with our trunks. We are using 4 channels of the PRI (channels 1-4), plus the D-channel for signaling. The PRI is a U.S.-based T1. With the FXO ports, we are sharing 1 with a fax credit card machine, and the other one is shared with a different fax, coming off of the fax's phone port (so there is pretty much no way for it to ever see or feel anything fax-related). I've looked a bit at the asterisk/full and messages log, but so far nothing
Re: [asterisk-users] resample not recognized by menuselect on Lenny
On Wed, Nov 26, 2008 at 08:33:25PM +0100, Philipp Kempgen wrote: BTW (sorry for hijacking the thread): What package satisfies the dependency on resample on Debian Lenny? It's not yet packaged. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] spandsp not recognized by menuselect on Lenny
On Wed, Nov 26, 2008 at 08:12:51PM +0100, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Wed, Nov 26, 2008 at 07:16:05PM +0100, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote: I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny. I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 : apt-get install libspandsp1 cd /usr/src/asterisk-1.6.0.1 ./configure make clean make menuselect In menuselect/application menu, I can see that app_fax is greyed out and cannot be selected. 1. My understanding is that both spandsp and libtiff4 are not daemons and do not need any configuration, right ? 2. How can I check spandsp is correctly installed and should be made available by menuselect ? Note: with apt-get install libspandsp1 on Lenny, spandsp-0.0.5pre4-1 and libtiff4 are installed and configured. aptitude install libspandsp-dev Yup. libspandsp is a build-dependency (in contrast to a normal runtime dependency). Thus you need the -dev package. Up-to-date list of build dependencies: http://svn.debian.org/viewsvn/pkg-voip/asterisk/branches/experimental/debian/control?rev=6480view=markup BTW: That list doesn't include libspandsp* :-) Oops -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Asterisk daemon dies about once per day
On Wed, Nov 26, 2008 at 12:59:49PM -0700, Douglas Mortensen wrote: OK. I know it's been a few weeks since my original post. Things have been busy ;-) Based on help from the trixbox forums and the asterisk-users mailing list, I have located 30 asterisk core dump files in /tmp. These date from 10/30/08 to 10/24/08. Today is 10/26/08. So this does agree with the intermittent nature of the problem. Many days there are no dumps. Other days there are 5, 7, 6, 1, 4, or 2 dumps. What's the bug number? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] 2 Asterisks to one PBX - E1 conection
Hi all, I have a question regarding connection of two Asterisk servers to our PBX. Each Asterisk server has one PCI E1 card, and they are in failover mode with Linux HA. On our PBX we have only one E1 card towards Asterisk servers. My question is how to connect these two Asterisks to one E1 card on PBX, and when primary Asterisk server fails not to have to manually pull out E1 cable from primary server and plug it in secondary server in order to have active connection to E1 card on PBX. Is there some kind of splitter which, on one side can accept two E1 connections from Asterisks and on the other side one E1 link from PBX. This splitter must also recognize towards which one of two E1 links on Asterisk side it should send signals to. eg. when primary Asterisk fails this splitter should send signals to its eg. port 2 (connection towards secondary Asterisk). I would be most grateful if someone could provide me with a link to such products. Thanks Dubravko ___ -- 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] 2 Asterisks to one PBX - E1 conection
Redfone On Wed, Nov 26, 2008 at 4:04 PM, dubravko caric [EMAIL PROTECTED] wrote: Hi all, I have a question regarding connection of two Asterisk servers to our PBX. Each Asterisk server has one PCI E1 card, and they are in failover mode with Linux HA. On our PBX we have only one E1 card towards Asterisk servers. My question is how to connect these two Asterisks to one E1 card on PBX, and when primary Asterisk server fails not to have to manually pull out E1 cable from primary server and plug it in secondary server in order to have active connection to E1 card on PBX. Is there some kind of splitter which, on one side can accept two E1 connections from Asterisks and on the other side one E1 link from PBX. This splitter must also recognize towards which one of two E1 links on Asterisk side it should send signals to. eg. when primary Asterisk fails this splitter should send signals to its eg. port 2 (connection towards secondary Asterisk). I would be most grateful if someone could provide me with a link to such products. Thanks Dubravko ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Softphone IP publico x privado
Pessoal, Me ocorreu uma dúvida: Imagine que tenho uma rede com um IP válido e um router compartilhando essa internet para 3 micros. Eu gostaria de colocar 3 ramais softphone nessas 3 máquinas cujo servidor fica fora da rede. Como fica a configuração do meu roteador para que isso funcione direito??? Obrigado, Luis A P Barbosa (61) 8482-2016 ___ -- 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] Dahdi, b410p and looping from 1 port to another
Olivier wrote: Doc says it should be turned on in those instances where a BRI is daisy-chained and terminated on the B410P in NT mode. So after reading this again, I would I should have turned it on. Do you agree ? B410P card1 - port1 - TE mode -- B410P card2 - port1 - NT mode -- B410P card2 - port2 - NT mode Your diagram does not make any sense. You show three ports being connected together, with two of them in NT mode. This is not possible, only one device on a multipoint BRI can be NT, the rest must be TE. Yes, if you really are connecting ports this way, the port in NT mode (really one of the ports, but it's easiest to just use the NT mode port) should be set to terminate the line. However, chan_dahdi + wcb4xxp + libpri do not currently support NT point-to-multipoint mode anyway, so this configuration cannot work with the current code. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] resample not recognized by menuselect on Lenny
Tzafrir Cohen wrote: On Wed, Nov 26, 2008 at 08:33:25PM +0100, Philipp Kempgen wrote: BTW (sorry for hijacking the thread): What package satisfies the dependency on resample on Debian Lenny? It's not yet packaged. Right, 'it' is Digium's redistribution of some resampling code, and it is available here: http://svn.digium.com/svn/thirdparty/libresample/trunk -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] CDR Hangupcause
Hi, I'm trying to get HANGUPCAUSE on my cdr the problem I'm facing is that this option: endbeforehexten=yes is not working at least on asterisk 1.6.0.1, so if I put yes o no I cant set CDR value with that value. It seems to finish the CDR record before h is executed. I'm using cdr_mysql. Any idea?? Thanks!! ___ -- 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] resample not recognized by menuselect on Lenny
Kevin P. Fleming schrieb: Tzafrir Cohen wrote: On Wed, Nov 26, 2008 at 08:33:25PM +0100, Philipp Kempgen wrote: BTW (sorry for hijacking the thread): What package satisfies the dependency on resample on Debian Lenny? It's not yet packaged. Right, 'it' is Digium's redistribution of some resampling code, and it is available here: http://svn.digium.com/svn/thirdparty/libresample/trunk Thanks for the pointer Kevin. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Dahdi, b410p and looping from 1 port to another
2008/11/26 Kevin P. Fleming [EMAIL PROTECTED] Olivier wrote: Doc says it should be turned on in those instances where a BRI is daisy-chained and terminated on the B410P in NT mode. So after reading this again, I would I should have turned it on. Do you agree ? B410P card1 - port1 - TE mode -- B410P card2 - port1 - NT mode -- B410P card2 - port2 - NT mode Your diagram does not make any sense. Sorry for the diagram : a false move made me send my previous email before I could finish it. Really sorry about that ! ___ -- 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] Dahdi, b410p and looping from 1 port to another
2008/11/26 Kevin P. Fleming [EMAIL PROTECTED] snip However, chan_dahdi + wcb4xxp + libpri do not currently support NT point-to-multipoint mode anyway, so this configuration cannot work with the current code. My (original) question was : Shall I turn NT 100 ohm termination when directly connecting one (NT) port to one TE port from a single B410P card ? Documentation speaks about daisy chains but I wonder if this applies to this case. regards ___ -- 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] Dahdi, b410p and looping from 1 port to another
Olivier wrote: normally if there are 2 devices you want termination on on both, when there are more than 2 in a chain, the one at each end gets terminated, not the middle ones. if its really a star, not a chain electrically - experiment a bit depends on the lengths of each arm what is best. on real short lengths of wire (all in the same room) probably won't make a difference either way. 2008/11/26 Kevin P. Fleming [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] snip However, chan_dahdi + wcb4xxp + libpri do not currently support NT point-to-multipoint mode anyway, so this configuration cannot work with the current code. My (original) question was : Shall I turn NT 100 ohm termination when directly connecting one (NT) port to one TE port from a single B410P card ? Documentation speaks about daisy chains but I wonder if this applies to this case. regards ___ -- 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] Dahdi, b410p and looping from 1 port to another
Jon Pounder wrote: on real short lengths of wire (all in the same room) probably won't make a difference either way. Agreed. All our lab testing of cards (short cables) does not involve enabling termination and it works fine. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Asterisk daemon dies about once per day
On Wednesday 26 November 2008 13:59:49 Douglas Mortensen wrote: OK. I know it's been a few weeks since my original post. Things have been busy ;-) Based on help from the trixbox forums and the asterisk-users mailing list, I have located 30 asterisk core dump files in /tmp. These date from 10/30/08 to 10/24/08. Today is 10/26/08. So this does agree with the intermittent nature of the problem. Many days there are no dumps. Other days there are 5, 7, 6, 1, 4, or 2 dumps. I have used the viewcore tool as indicated on http://www.voip-info.org/wiki-Asterisk+debugging on one of the most recent dump files and posted the output here: http://kgotsi.com/static.php?page=static-asterisk-core-dumps I really don't know what all of the output it produced means, so I'm relying on others with more expertise here to take a look and tell me what insight it may provide. I'm not sure who had the idea on running viewcore, but that information is about the most useless debug output I've seen. I'd recommend that you follow the steps with gdb, as gdb provides FAR more useful output. Also, Fonality binaries are not supported here. You'd have to go to them directly for any support. FYI, I have not recompiled asterisk yet. I'm a little nervous about downtime that could be caused by the process not going smoothly, but definitely willing to do so if that's what is needed to fix this problem. Recompiling with the steps outlined in Murf's email is definitely the way to go to get a resolution to your problem from this forum. -- Tilghman ___ -- 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] Asterisk daemon dies about once per day
On Wed, Nov 26, 2008 at 04:20:44PM -0600, Tilghman Lesher wrote: On Wednesday 26 November 2008 13:59:49 Douglas Mortensen wrote: OK. I know it's been a few weeks since my original post. Things have been busy ;-) Based on help from the trixbox forums and the asterisk-users mailing list, I have located 30 asterisk core dump files in /tmp. These date from 10/30/08 to 10/24/08. Today is 10/26/08. So this does agree with the intermittent nature of the problem. Many days there are no dumps. Other days there are 5, 7, 6, 1, 4, or 2 dumps. I have used the viewcore tool as indicated on http://www.voip-info.org/wiki-Asterisk+debugging on one of the most recent dump files and posted the output here: http://kgotsi.com/static.php?page=static-asterisk-core-dumps I really don't know what all of the output it produced means, so I'm relying on others with more expertise here to take a look and tell me what insight it may provide. I'm not sure who had the idea on running viewcore, but that information is about the most useless debug output I've seen. I'd recommend that you follow the steps with gdb, as gdb provides FAR more useful output. Read: install the packages gdb and asterisk-dbg (of the same version as asterisk. *don't* upgrade). Then you can get useful backtraces. Also, Fonality binaries are not supported here. You'd have to go to them directly for any support. Trixbox CE's sources are publically available and rebuildable. Source RPMs are available at http://yum.trixbox.org/centos/5/SRPMS/ FYI, I have not recompiled asterisk yet. I'm a little nervous about downtime that could be caused by the process not going smoothly, but definitely willing to do so if that's what is needed to fix this problem. Recompiling with the steps outlined in Murf's email is definitely the way to go to get a resolution to your problem from this forum. Once you follow that, you don't necessarily have the original system. Let's first extract data before starting to change the examined system. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] 2 Asterisks to one PBX - E1 conection
dubravko caric wrote: Hi all, I have a question regarding connection of two Asterisk servers to our PBX. Each Asterisk server has one PCI E1 card, and they are in failover mode with Linux HA. On our PBX we have only one E1 card towards Asterisk servers. My question is how to connect these two Asterisks to one E1 card on PBX, and when primary Asterisk server fails not to have to manually pull out E1 cable from primary server and plug it in secondary server in order to have active connection to E1 card on PBX. Is there some kind of splitter which, on one side can accept two E1 connections from Asterisks and on the other side one E1 link from PBX. This splitter must also recognize towards which one of two E1 links on Asterisk side it should send signals to. eg. when primary Asterisk fails this splitter should send signals to its eg. port 2 (connection towards secondary Asterisk). I would be most grateful if someone could provide me with a link to such products. Thanks Dubravko Don't know how well it works, but we've been looking at these: http://www.rhinoequipment.com/1portfail.html Alex ___ -- 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] 2 Asterisks to one PBX - E1 conection
Is there some kind of splitter which, on one side can accept two E1 connections from Asterisks and on the other side one E1 link from PBX. This splitter must also recognize towards which one of two E1 links on Asterisk side it should send signals to. eg. when primary Asterisk fails this splitter should send signals to its eg. port 2 (connection towards secondary Asterisk). We bought this, and loaded it up with 8 T1 ports http://www.wti.com/index.php?page=shop.product_detailscategory_id=7flypage=flypage-ask.tplproduct_id=56option=com_virtuemartItemid=33 The AFS-16 has a serial port, which accepts commands to toggle some, or all ports. If you get yourself a startech.com netrs2321 sold at several places on the internet, including directly from startech.com which puts the serial port behind a tcp service on ethernet, and lets any of several machines manage the serial on the AFS 16 box. We wrote a Perl script, put it on both systems, and then when the script detects a failure it flips the AFS 16 ports, assuming it doesn't already have the lines. So this is a solution that allows for total failure of one of the boxes, in fact, the way to test it out is to yank an ethernet cable, ping fails, and then the script flips the switch ports. Now with all that said, I'm not sure how the prices for this solution compare to the other solutions proposed on this thread. Get your own quote, but the AFS-16 box was a lot of money for what it does. And now that I'm about to scale out of this setup, I think it's better to instead terminate the T1s into a box that converts them to VoIP traffic, and do any failover to the Asterisk boxes using DNS or other TCP/IP tricks. I suppose you could say that just moves the single point of failure to the gear that does the T1-to-VoIP conversion, probably a Cisco box. I think the Cisco box is a better long-term investment, as it scales, trades-up, and can be used for lots of other things, or even resold when your needs change. The other better thing about that approach is that your Asterisk systems speak VoIP, so if you get into a situation where you can find a VoIP service provider that gets better pricing than your regular phone provider (so far we haven't, once you add in the bandwidth costs) you're ready for that day. You could also keep all the gear you have, but change the way it's laid out. Terminate the Ts into the Asterisk cards, but have a system just do the T-to-voip conversion. I'd rather have an appliance do that, and there are probably some other appliances than the Cisco boxes. ___ -- 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] pick up IAX2 calls
Somebody know some work around for it? I still trying to find a solution but nothing seems to work thanks Eric ManxPower Wieling wrote: The problem is that IAX2 does not seem to support call pickup. Bruno Castelo Branco wrote: hi I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 for all IAX extensions in iax.conf. Didn't works for while. thanks Tim Panton wrote: I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I think that if you have a Dial() that rings several extens, then any of the technologies involved can pickup with *8 So if you have Dial(IAX/fredSIP/billzap/mark) then someone in the same group as fred can pickup with IAX and someone in the same group as bill can pickup with SIP etc. So it's an asterisk thing, not an IAX thing per-se. Tim. P.S. (you could try putting in a dummy 'fred' entry into Dial and iax.conf.) T. On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote: hi thanks Luis , but doesn't work. For SIP extensions works well *8, but for IAX a tried *8 and ** + iax extension and didn't works Luis Morales wrote: Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- 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] Ring/Off-hook in strange state 6 channel X
Versions - Asterisk 1.4.22 - DAHDI Linux 2.0.0 - DAHDI Tools 2.0.0 - Libpri 1.4.7 - Addons 1.4.7 Here is chan_dahdi.conf ; ; DAHDI telephony interface [trunkgroups] [channels] context=from-pstn switchtype=national signalling=fxo_ks rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes immediate=no busydetect=no callprogress=no answeronpolarityswitch=yes hanguponpolarityswitch=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 #include dahdi-channels.conf File: dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Mon Nov 24 16:19:00 2008 -- do not hand edit ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER) ;;; line=3 WCTDM/0/2 FXSKS (EC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 3 callerid= group= context=default ; Span 2: XBUS-00/XPD-00 Xorcom XPD #00/00: FXO ;;; line=5 XPP_FXO/00/00/0 signalling=fxs_ks callerid=asreceived cidsignalling=v23 cidstart=polarity callerid=asreceived callwaiting=no group=0 context=from-pstn channel = 5-12 Greetings List I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all of them give me the error Ring/Off-hook in strange state 6. Whenever the caller hangup, the call continue to execute until it hits the hard coded hangup. I changed chan_dadhi busydetect=no and callprogress=no but problem still persist. I also tried to use different PABX in vain. GSM modem (FUSION100) also produces no useful result Please help Sam Muro ___ -- 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] Softphones with RPID and BLF
Hello, I am looking for a softphone which supports RPID (displaying the called party name) and BLF features. I couldn't find one so far... Any idea whether such a softphone exists? Thanks! __Yehavi: ___ -- 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] pick up IAX2 calls
Hello
I asked the same thing some time ago, but nobody answered.
I founded some workaround.
Use this in your dialplan:
exten = _7.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:1})
exten = _7.,n,Pickup(${EXTEN:[EMAIL PROTECTED])
This worked for me.
Cosmin
--- On Thu, 11/27/08, Bruno Castelo Branco [EMAIL PROTECTED] wrote:
From: Bruno Castelo Branco [EMAIL PROTECTED]
Subject: Re: [asterisk-users] pick up IAX2 calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thursday, November 27, 2008, 4:59 AM
Somebody know some work around for it?
I still trying to find a solution but nothing seems to work
thanks
Eric ManxPower Wieling wrote:
The problem is that IAX2 does not seem to support call pickup.
Bruno Castelo Branco wrote:
hi
I'm using only IAX extensions and inserted callgroup=1 and callpickup=1
for all IAX extensions in iax.conf. Didn't works for while.
thanks
Tim Panton wrote:
I think it doesn't work across channel types.
So it works (if I recall correctly) in IAX or in SIP or in ZAP,
but not in mixture.
I think that if you have a Dial() that rings several extens,
then any of the technologies involved can pickup with *8
So if you have Dial(IAX/fredSIP/billzap/mark)
then someone in the same group as fred can pickup with IAX
and someone in the same group as bill can pickup with SIP
etc.
So it's an asterisk thing, not an IAX thing per-se.
Tim.
P.S.
(you could try putting in a dummy 'fred' entry into Dial and iax.conf.)
T.
On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote:
hi
thanks Luis , but doesn't work.
For SIP extensions works well *8, but for IAX a tried *8 and ** + iax
extension and didn't works
Luis Morales wrote:
Try with ** + iax extension
Regards,
Luis Morales
On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hi
Somebody knows if pickup call works with IAX2?
I enable *8 in features.conf, but doesn't works with IAX2 extensions.
Any idea?
thanks
___
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Re: [asterisk-users] 2 Asterisks to one PBX - E1 conection
Hi Steve, yes I know about RedFone, in fact I'm already using it on three locations. now I'm looking for similar solution but with PCI cards. Thanks /davor From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 26, 2008 10:11:00 PM Subject: Re: [asterisk-users] 2 Asterisks to one PBX - E1 conection Redfone On Wed, Nov 26, 2008 at 4:04 PM, dubravko caric [EMAIL PROTECTED] wrote: Hi all, I have a question regarding connection of two Asterisk servers to our PBX. Each Asterisk server has one PCI E1 card, and they are in failover mode with Linux HA. On our PBX we have only one E1 card towards Asterisk servers. My question is how to connect these two Asterisks to one E1 card on PBX, and when primary Asterisk server fails not to have to manually pull out E1 cable from primary server and plug it in secondary server in order to have active connection to E1 card on PBX. Is there some kind of splitter which, on one side can accept two E1 connections from Asterisks and on the other side one E1 link from PBX. This splitter must also recognize towards which one of two E1 links on Asterisk side it should send signals to. eg. when primary Asterisk fails this splitter should send signals to its eg. port 2 (connection towards secondary Asterisk). I would be most grateful if someone could provide me with a link to such products. Thanks Dubravko ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
