Re: [asterisk-users] No tone, one way communcation.
On Mon, Oct 26, 2009 at 11:20:07PM -0700, PATRICK KANGETHE wrote: > my lsdahdi output is; > > 1. [r...@elastix ~]# lsdahdi > ### Span 1: WCTDM/8 "YSTDM8xx REV E Board 9" (MASTER) > 1 FXSFXOKS (In use) > 2 FXSFXOKS (In use) > 3 EMPTY > 4 FXSFXOKS (In use) > 5 FXOFXSKS (In use) RED > 6 FXOFXSKS (In use) RED > 7 FXOFXSKS (In use) RED > 8 FXOFXSKS (In use) RED > > seems fxs port 3 is disconnected but i will fix it. The analog phone is > connected to fxs port 1. What is the output of: asterisk -rx 'dahdi show channels' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI not detecting RINGING Status on the Channel
On Mon, Oct 26, 2009 at 09:02:10PM -0300, Mariano Lecuona wrote: > For some reason I am not able to set loopstart instead of kewlstart: > > Console out put: > > [Oct 26 20:58:40] == Parsing '/etc/asterisk/chan_dahdi.conf': [Oct 26 > 20:58:40] Found > [Oct 26 20:58:40] -- Registered channel 1, FXS Loopstart signalling > [Oct 26 20:58:40] -- Registered channel 2, FXS Loopstart signalling > [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:7677 mkintf: Signalling > requested on channel 3 is FXS Loopstart but line is in FXS Kewlstart > signalling > [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:11294 build_channels: Unable to > register channel '1-8' > pbx*CLI> module load chan_dahdi.so > > any ideas? What is the output of lsdahdi ? Have you edited /etc/dahdi/system.conf ? To apply changes there, run dahdi_cfg . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing Asterisk
hi , i have started reading asterisk book need your guidance.friend as i am newbie in asterisk so plz plz forgive me if i ask stupid questions. INSTALLING ASTERISK - on which linux flavour i should start the installation of asterisk (CentOs,Fedora,Ubuntu) right now i am using ubuntu 8.04lts - What are the essential packages. - whats is PRI, BRI interfaces - what is asterisk now Regards, Pawan___ -- 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] Installing Asterisk
- on which linux flavour i should start the installation of asterisk (CentOs,Fedora,Ubuntu) right now i am using ubuntu 8.04lts I use Centos 5.3 - What are the essential packages. The book explains which packages you need to get a basic Asterisk up and running. - whats is PRI, BRI interfaces These are used if you want to connect Asterisk to normal standard phone lines. If you are just using SIP and making internet calls, you don’t need these. You can call to and from standard phone lines if you get a SIP provider. Take a look at this website. It has quite a good list of providers:- http://www.voip-info.org/wiki/view/DID+Service+Providers - what is asterisk now It is an installation DVD containing all of the basic options needed to run asterisk. If you use it, then you wont need to go through most of the configuration that the book discusses. I preferred to use the book, because then i know a bit more about what is happening. Dan Journo From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of aster...@opensourcesolution.in Sent: 27 October 2009 09:34 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Installing Asterisk hi , i have started reading asterisk book need your guidance.friend as i am newbie in asterisk so plz plz forgive me if i ask stupid questions. Installing Asterisk - on which linux flavour i should start the installation of asterisk (CentOs,Fedora,Ubuntu) right now i am using ubuntu 8.04lts - What are the essential packages. - whats is PRI, BRI interfaces - what is asterisk now Regards, Pawan ___ -- 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] installing
installing asterisk___ -- 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] installing
On 27 Oct 2009, at 09:49, wrote: > installing asterisk Me too! ___ -- 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] installing
aster...@opensourcesolution.in wrote: > installing asterisk I am intrigued by your ideas and would like to subscribe to your quarterly newsletter, as well as attend your biannual leadership seminar. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] [OT] Snom M9
Hi, Does anybody know when the M9 is actually being launched ? All I have read is late October. Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] RTP timestamps
Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should be, but sometimes while the asterisk plays MOH (or somebody transfers call to another extension) the timestamps on RTP packets will fall to past. Providers media gateway dosn't like that. The marker bit is correctly set but it seems like that dosn't change anything. Sequences and SSRC-s are Ok, no packet loss. Has anyone seen something like this before and knows what is the cause and how to fix this? I've tried many changes in config and upgraded to 1.6.1 but it didnt change anything, currently running asterisk 1.4.26.1 on 64 bit intel platform with opensuse. Here is the tcpdump view from wireshark, xxx is providers ip and yyy is asterisk: 6218207.717454xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54364, Time=1987711680 6219207.717481yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22826, Time=2202453496 6220207.737442xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54365, Time=1987711840 6221207.757430xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54366, Time=1987712000 6222207.759283yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22827, Time=736089280, Mark 6223207.765349yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22828, Time=736089440 Help! Greetings, Liivo ___ -- 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] Installing Asterisk
Go to sites like digium.com, asterisk.org, asteriskguru.com, trixbox.com,elastix.org for more understanding. Goodluck. From: "aster...@opensourcesolution.in" To: asterisk-users@lists.digium.com Sent: Tue, October 27, 2009 12:34:18 PM Subject: [asterisk-users] Installing Asterisk hi , i have started reading asterisk book need your guidance.friend as i am newbie in asterisk so plz plz forgive me if i ask stupid questions. Installing Asterisk - on which linux flavour i should start the installation of asterisk (CentOs,Fedora,Ubuntu) right now i am using ubuntu 8.04lts - What are the essential packages. - whats is PRI, BRI interfaces - what is asterisk now Regards, Pawan ___ -- 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] Fax with a AEx410P and Beronet BN4S0 => Sending Problem
Hi all, I have problem with one of my configuration : FAX <-> AEX410P (One FXS port) <---> BN4S0 <> PSTN Case 1 : Receiving Fax is Ok ( PSTN ---> BN4SO --> AEX410P --> FAX ) Case 2 : Sending Fax is nok ( FAX ---> AEX410P --> BN4SO --> PSTN ) I think we have some synchronisation problem because , de beginning of the fax is correct but I have some blank line on document and Asterisk do not release the line. This is the result of dahdi show channel 1 : Channel: 1LI> File Descriptor: 13 Span: 1 Extension: Dialing: no Context: from-internal Caller ID: 70 Calling TON: 0 Caller ID name: device Mailbox: 7...@device Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook And this is configration of midn port : [PORT 2] -> name: swisscom -> allowed_bearers: all -> far_alerting: no -> rxgain: 0 -> txgain: 0-> te_choose_channel: no -> pmp_l1_check: no -> reject_cause: 21 -> block_on_alarm: no -> hdlc: no -> context: ext-did-0002-> language: en -> musicclass: default -> callerid: -> method: standard -> dialplan: 0 -> localdialplan: 0 -> cpndialplan: 0 -> nationalprefix: 0-> internationalprefix: 00 -> presentation: -1 -> screen: -1 -> always_immediate: no -> nodialtone: no -> immediate: no-> senddtmf: no -> astdtmf: no -> hold_allowed: no -> early_bconnect: yes -> incoming_early_audio: no -> echocancel: 128 -> need_more_infos: no -> noautorespond_on_setup: no -> nttimeout: no -> bridging: yes-> jitterbuffer: 4000 -> jitterbuffer_upper_threshold: 0 -> callgroup: -> pickupgroup: -> max_incoming: -1 -> max_outgoing: -1 -> l1watcher_timeout: 0 -> overlapdial: 0 -> msns: * -> faxdetect: no-> faxdetect_context: -> faxdetect_timeout: 5 -> ptp: no Somebody already have this problem ? Thanks for your precious help, Vincent ___ -- 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] hangup from which side
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
On Mon, Oct 26, 2009 at 9:05 AM, Danny Nicholas wrote:
> So this *should* work??
> [outgoing]
> - exten => s,1,Dial(DAHDI/1/5551212,20)
> - exten => s,2,Noop(I hung up)
> - exten => s,3,Hangup
> - exten => h,1,Noop(you hung up)
> - exten => h,2,Hangup
>
> [incoming]
> - exten => s,1,Answer
> - exten => s,2,Noop(I hung up)
> - exten => s,3,Hangup
> - exten => h,1,noop(you hung up)
> - exten => h,2,hangup
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
> Sent: Friday, October 23, 2009 1:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] hangup from which side
>
> if you are debugging visually then look at SIP BYE message ... who sent it
> first
> and on PRI who sent the DISCONNECT message first.
>
> if you need to know that in the dialplan ... then if the originating
> channel hanged up
> then the dialplan should stop executing and go straight to h,1 even if
> Dial(,,g) is used
>
> also there is a channel variable HANGUPCAUSE and you can check what it
> does on the next step
> with Dial(,,g) and on h,1 ... since I don't know :)
>
> Martin
>
> On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH wrote:
>> When Asterisk establish a call through an outbound trunk, Is there any way
> I
>> can know who hang up the call first? The caller or the party called?
>>
>>
>>
>> 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
Re: [asterisk-users] hangup from which side
That will work on an outgoing call. Apparently (AFAICS) there is no feature
in Answer to jump to H or continue like the Dial command has.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Tuesday, October 27, 2009 8:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
On Mon, Oct 26, 2009 at 9:05 AM, Danny Nicholas wrote:
> So this *should* work??
> [outgoing]
> - exten => s,1,Dial(DAHDI/1/5551212,20)
> - exten => s,2,Noop(I hung up)
> - exten => s,3,Hangup
> - exten => h,1,Noop(you hung up)
> - exten => h,2,Hangup
>
> [incoming]
> - exten => s,1,Answer
> - exten => s,2,Noop(I hung up)
> - exten => s,3,Hangup
> - exten => h,1,noop(you hung up)
> - exten => h,2,hangup
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
> Sent: Friday, October 23, 2009 1:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] hangup from which side
>
> if you are debugging visually then look at SIP BYE message ... who sent it
> first
> and on PRI who sent the DISCONNECT message first.
>
> if you need to know that in the dialplan ... then if the originating
> channel hanged up
> then the dialplan should stop executing and go straight to h,1 even if
> Dial(,,g) is used
>
> also there is a channel variable HANGUPCAUSE and you can check what it
> does on the next step
> with Dial(,,g) and on h,1 ... since I don't know :)
>
> Martin
>
> On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH
wrote:
>> When Asterisk establish a call through an outbound trunk, Is there any
way
> I
>> can know who hang up the call first? The caller or the party called?
>>
>>
>>
>> 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] RTP timestamps
Liivo, I wonder if you are dealing with this general class of issues: https://issues.asterisk.org/view.php?id=11491 -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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 not detecting RINGING Status on the Channel
I have plugges only 2 lines. That's why the rest is in RED [r...@pbx ~]# lsdahdi ### Span 1: WCTDM/8 "YSTDM8xx REV E Board 9" (MASTER) 1 FXOFXSLS (In use) 2 FXOFXSLS (In use) 3 FXOFXSKS (In use) RED 4 FXOFXSKS (In use) RED 5 FXOFXSKS (In use) RED 6 FXOFXSKS (In use) RED 7 FXOFXSKS (In use) RED 8 FXOFXSKS (In use) RED 2009/10/27 Tzafrir Cohen > On Mon, Oct 26, 2009 at 09:02:10PM -0300, Mariano Lecuona wrote: > > For some reason I am not able to set loopstart instead of kewlstart: > > > > Console out put: > > > > [Oct 26 20:58:40] == Parsing '/etc/asterisk/chan_dahdi.conf': [Oct 26 > > 20:58:40] Found > > [Oct 26 20:58:40] -- Registered channel 1, FXS Loopstart signalling > > [Oct 26 20:58:40] -- Registered channel 2, FXS Loopstart signalling > > [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:7677 mkintf: Signalling > > requested on channel 3 is FXS Loopstart but line is in FXS Kewlstart > > signalling > > [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:11294 build_channels: Unable > to > > register channel '1-8' > > pbx*CLI> module load chan_dahdi.so > > > > any ideas? > > What is the output of lsdahdi ? > > Have you edited /etc/dahdi/system.conf ? To apply changes there, run > dahdi_cfg . > > -- > Tzafrir Cohen > icq#16849755 > jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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] Installing Asterisk
Dan Journo wrote: > > - on which linux flavour i should start the installation of asterisk > (CentOs,Fedora,Ubuntu) > right now i am using ubuntu 8.04lts > > I use Centos 5.3 > CentOS 5.3 works well, though this is somewhat of a religious argument If you are familiar with Ubadoobie, then why not use it? I was never successful with an earlier version, but that is me. fedora seems to have a short life, and many don't consider it a wise choice. > - What are the essential packages. > > The book explains which packages you need to get a basic Asterisk up > and running. > there are also several good web sites that tick off all the packages needed Google can help with this > > > - whats is PRI, BRI interfaces > > These are used if you want to connect Asterisk to normal standard > phone lines. If you are just using SIP and making internet calls, you > don’t need these. You can call to and from standard phone lines if you > get a SIP provider. > > Take a look at this website. It has quite a good list of providers:- > > http://www.voip-info.org/wiki/view/DID+Service+Providers > > > - what is asterisk now > > It is an installation DVD containing all of the basic options needed > to run asterisk. If you use it, then you wont need to go through most > of the configuration that the book discusses. > > I preferred to use the book, because then i know a bit more about what > is happening. > I agree, though depending on your goal, it may be the best choice. If one wants to get up and running quickly, and is willing to have a tougher time when something doesn't work correctly, and is willing use the defaults, then it may be the better choice. If, however, one wants a learning experience, takes notes, and follow the now somewhat outdated TFOT book, then go the install from scratch route. Best of luck John Novack > *From:* asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of > *aster...@opensourcesolution.in > *Sent:* 27 October 2009 09:34 > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] Installing Asterisk > > hi , > > i have started reading asterisk book need your guidance.friend as i am > newbie in asterisk so plz plz forgive me if i ask stupid questions. > > > *Installing Asterisk* > > - on which linux flavour i should start the installation of asterisk > (CentOs,Fedora,Ubuntu) > right now i am using ubuntu 8.04lts > > - What are the essential packages. > > - whats is PRI, BRI interfaces > > - what is asterisk now > > Regards, > > Pawan > > > > ___ > -- 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 > > > > > Checked by AVG - www.avg.com > Version: 8.5.423 / Virus Database: 270.14.33/2461 - Release Date: 10/26/09 > 20:22:00 > > -- Dog is my co-pilot ___ -- 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] pri intense debug span 1
I posted a message the other day http://lists.digium.com/pipermail/asterisk-users/2009-October/239728.html about not being able to use newer versions of libpri. I am stuck at libpri 1.4.1 I posted the "pri intense debug span 1" output for a good call and failed call. Was wondering what I can do next to give the list information to help find out what the issue is on the PRI. Thanks, 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] pri intense debug span 1
While this list provides VERY good information, it should not be your only course of action. You should be looking at the Digium forums, googling and opening a ticket and/or browsing the issue list. Also check asterisk-guru and voip-info.org for further information. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Tuesday, October 27, 2009 9:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] pri intense debug span 1 I posted a message the other day http://lists.digium.com/pipermail/asterisk-users/2009-October/239728.html about not being able to use newer versions of libpri. I am stuck at libpri 1.4.1 I posted the "pri intense debug span 1" output for a good call and failed call. Was wondering what I can do next to give the list information to help find out what the issue is on the PRI. Thanks, 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to dial multiple extensions at once like in a ring group and put them in conference?
Hi, I have to set this up for a client, where he could dial multiple extensions at once, and then put all who picks up into a conference. I am using a script which does it using originate command. But the originate commands run one after another, and so it takes a few seconds to call the extensions, one after another. This is not acceptable by the client. If I use Dial(SIP/201&SIP/202..., it is like a ringgroup, where if one extension picks up, others stop ringing. Is there a way to dial all the extensions at once and then put them in the conference? -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?
Use an AGI that does a Mass originate/call to ring everyone at once. Have the AGI do an originate loop using a context to dump into the conference and call it from the dialplan like this: - exten => s,1,AGI(massconf.agi|ext1|ext2|ext3|ext4|ext5.) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, October 27, 2009 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference? Hi, I have to set this up for a client, where he could dial multiple extensions at once, and then put all who picks up into a conference. I am using a script which does it using originate command. But the originate commands run one after another, and so it takes a few seconds to call the extensions, one after another. This is not acceptable by the client. If I use Dial(SIP/201&SIP/202..., it is like a ringgroup, where if one extension picks up, others stop ringing. Is there a way to dial all the extensions at once and then put them in the conference? -- Zeeshan A Zakaria ___ -- 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 interconnection problem
Someone? As * is used so extensively with SIP I must've made a _glaring_
mistake in my config (!)
/Rob
Robert Bielik skrev:
> Tarek Sawah skrev:
>> you need to post you SIP.conf and your Extensions.conf so someone can
>> have a look at them and see if there is anything missing
>> what are the contexts you are using with your peers?
>> what is the dial plan triggered when calling your destination number?
>
> Machine 1 ---
> iax.conf: ==
> [general]
> bandwidth=low
> disallow=lpc10 ; Icky sound quality... Mr. Roboto.
> jitterbuffer=no
> forcejitterbuffer=no
> autokill=yes
>
> [2200]
> type=friend
> host=dynamic
> context=users
> username=2200
> secret=none
> auth=md5
>
> sip.conf ===
> [general]
> port=5060
> bindaddr=0.0.0.0
>
> disallow=all
> allow=alaw ; Allow codecs in order of preference
> allow=ulaw
> allow=gsm
> allow=g726
>
> dtmfmode=rfc2833
>
> register => machine_1:wabo...@192.168.10.77/machine_2
>
> [machine_2]
> allow=alaw,ulaw,gsm,g726
> host=dynamic
> secret=wabooba
> type=friend
> context=sip_incoming
> username=machine_2
>
> extensions.conf ==
> [general]
> static=yes
> writeprotect=no
> clearglobalvars=no
>
> [globals]
> ; The outgoing sip trunk
> SIP_TRUNK=192.168.10.77
> OUTGOING_PREFIX=0
>
> [default]
> include => sip-incoming
> include => test
>
> [test]
> ; Create an extension, 600, for evaluating echo latency.
> ;
> exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
> exten => 600,n,Echo ; Do the echo test
> exten => 600,n,Playback(demo-echodone) ; Let them know it's over
> exten => 600,n,Goto(s,6)
>
> [users]
> include => sip-incoming
> include => outgoing
> include => test
>
> [sip-incoming]
> include => agi-async
> include => internal
>
> [agi-async]
> exten => _01,1,Agi(agi:async)
>
> [internal]
> exten => _2XXX,1,NoOp()
> exten => _2XXX,n,Dial(IAX2/${EXTEN})
> exten => _2XXX,n,Hangup()
>
> [outgoing-agi-async]
> exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${ext...@${sip_trunk})
> exten => _${OUTGOING_PREFIX}.,n,Set(CALLERID(name)=reason-${DIALSTATUS})
> exten => _${OUTGOING_PREFIX}.,n,Agi(agi:async)
>
> [outgoing]
> exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${SIP_TRUNK}/${EXTEN:1})
> exten => _${OUTGOING_PREFIX}.,n,Hangup()
>
> Machine 2
> sip.conf ===
> [general]
> port=5060
> bindaddr=0.0.0.0
>
> disallow=all
> allow=alaw ; Allow codecs in order of preference
> allow=ulaw
> allow=gsm
> allow=g726
>
> dtmfmode=rfc2833
>
> register => machine_2:wabo...@192.168.10.11/machine_1
>
> [machine_1]
> allow=alaw,ulaw,gsm,g726
> host=dynamic
> secret=wabooba
> type=friend
> context=sip_incoming
> username=machine_1
>
> extensions.conf ==
> [globals]
> ; The outgoing sip trunk
> SIP_TRUNK=192.168.10.11
>
> Rest is exactly the same. I have a zoiper connected to each machine and I'm
> trying to make a call from Machine 2 to zoiper
> on Machine 1:
>
> -- Registered IAX2 '2200' (AUTHENTICATED) at 192.168.10.113:4569
> -- Accepting AUTHENTICATED call from 192.168.10.113:
>> requested format = gsm,
>> requested prefs = (),
>> actual format = gsm,
>> host prefs = (),
>> priority = mine
> -- Executing [02...@users:1] Dial("IAX2/2200-1200",
> "SIP/192.168.10.11/2200") in new stack
> == Using SIP RTP CoS mark 5
> -- Called 192.168.10.11/2200
> [Oct 26 09:20:25] NOTICE[20248]: chan_sip.c:15031 handle_response_invite:
> Failed to authenticate on INVITE to '"2200"
> ;tag=as6173091f'
> -- SIP/192.168.10.11-090c2ea8 is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing [02...@users:2] Hangup("IAX2/2200-1200", "") in new stack
> == Spawn extension (users, 02200, 2) exited non-zero on 'IAX2/2200-1200'
> -- Hungup 'IAX2/2200-1200'
>
> Besides that "sip show peers" on either machine shows the other one correctly
> registered, and "iax2 show peers" shows the connected zoiper on each machine.
>
> Ideas, please ??
>
> TIA
> /Rob
>
>
>
> ___
> -- 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] installing
Lol Sent from my iPod On Oct 27, 2009, at 6:59 AM, Alex Balashov wrote: > aster...@opensourcesolution.in wrote: > >> installing asterisk > > I am intrigued by your ideas and would like to subscribe to your > quarterly newsletter, as well as attend your biannual leadership > seminar. > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > > ___ > -- 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] How to dial multiple extensions at once like in aring group and put them in conference?
Hi Danny, This is exactly what I am doing, but it takes a few seconds before all the extensions are ringing. The loop takes its time. I need something as quick as Dial(SIP/201&SIP/202... which is truly call all at once, but it connects only two channels, i.e. the first once which picked up, and then stops ringing the rest. Zeeshan On Tue, Oct 27, 2009 at 10:27 AM, Danny Nicholas wrote: > Use an AGI that does a Mass originate/call to ring everyone at once. > Have the AGI do an originate loop using a context to dump into the > conference and call it from the dialplan like this: > > - exten => s,1,AGI(massconf.agi|ext1|ext2|ext3|ext4|ext5…) > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria > *Sent:* Tuesday, October 27, 2009 9:19 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] How to dial multiple extensions at once like > in aring group and put them in conference? > > > > Hi, > > I have to set this up for a client, where he could dial multiple extensions > at once, and then put all who picks up into a conference. > > I am using a script which does it using originate command. But the > originate commands run one after another, and so it takes a few seconds to > call the extensions, one after another. This is not acceptable by the > client. > > If I use Dial(SIP/201&SIP/202..., it is like a ringgroup, where if one > extension picks up, others stop ringing. > > Is there a way to dial all the extensions at once and then put them in the > conference? > > -- > Zeeshan A Zakaria > > ___ > -- 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] How to dial multiple extensions at once like in aring group and put them in conference?
Have you tryed to generate .call files at once ? ___ -- 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] No tone, one way communcation.
Once the card was configured correctly, have you set on the GUI the correct port to your zap extension? On Mon, 26 Oct 2009 05:02:18 -0700 (PDT), PATRICK KANGETHE wrote: > 1. When i connected my analog phone to fxs card, i cannot get dial tone > what could be the problem? > > I am using elastix 1.5.2 based on centos 5.2 Final. > > 2. On my 2 sip softphones using x-lite linux versions, i get one way audio > how do i solve this?. This problem is also present when i use a windows > version on one end and linux version on other end. > > Any help will be highly appreciated. > > > > -- Atentamente, Jorge Gutiérrez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once likein aring group and put them in conference?
This might be a better application of a call file than an AMI originate. The AMI originate in this case has to operate in a threaded fashion, whereas if you created a call file for each extension and dumped them into /var/spool/asterisk/outgoing, pbx.c would call all of them at once without the "first pickup" problem. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, October 27, 2009 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to dial multiple extensions at once likein aring group and put them in conference? Hi Danny, This is exactly what I am doing, but it takes a few seconds before all the extensions are ringing. The loop takes its time. I need something as quick as Dial(SIP/201&SIP/202... which is truly call all at once, but it connects only two channels, i.e. the first once which picked up, and then stops ringing the rest. Zeeshan On Tue, Oct 27, 2009 at 10:27 AM, Danny Nicholas wrote: Use an AGI that does a Mass originate/call to ring everyone at once. Have the AGI do an originate loop using a context to dump into the conference and call it from the dialplan like this: - exten => s,1,AGI(massconf.agi|ext1|ext2|ext3|ext4|ext5.) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, October 27, 2009 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference? Hi, I have to set this up for a client, where he could dial multiple extensions at once, and then put all who picks up into a conference. I am using a script which does it using originate command. But the originate commands run one after another, and so it takes a few seconds to call the extensions, one after another. This is not acceptable by the client. If I use Dial(SIP/201&SIP/202..., it is like a ringgroup, where if one extension picks up, others stop ringing. Is there a way to dial all the extensions at once and then put them in the conference? -- Zeeshan A Zakaria ___ -- 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] RTP timestamps
Hi Alex, Yes, it's almost the same, except the fact that in my case timestamps sometimes decrease drastically. In internal network I have Snom 3xx phones with upgraded firmware, internal leg has no issues, i captured both legs and phones-asterisk part is ok, the other part, asterisk-provider has these issues which are mentioned above. Greetings, Liivo 27.10.2009 15:28, Alex Balashov kirjutas: > Liivo, > > I wonder if you are dealing with this general class of issues: > > https://issues.asterisk.org/view.php?id=11491 > > -- 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] How to dial multiple extensions at once like in aring group and put them in conference?
Zeeshan Zakaria escribió: > Hi Danny, > > This is exactly what I am doing, but it takes a few seconds before all > the extensions are ringing. The loop takes its time. > Are you originating the calls asynchronously? -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?
I think I should try the .call files. I haven't tried them in this particular scenario yet. Miguel, what exactly you mean by calling asynchronously? I do 'originate' in a 'while' loop once I have retrieved all the extensions to dial from the database. -- Zeeshan A Zakaria On Tue, Oct 27, 2009 at 11:07 AM, Miguel Molina wrote: > Zeeshan Zakaria escribió: > > Hi Danny, > > > > This is exactly what I am doing, but it takes a few seconds before all > > the extensions are ringing. The loop takes its time. > > > Are you originating the calls asynchronously? > > -- > Ing. Miguel Molina > Grupo de Tecnología > Millenium Phone Center > > > ___ > -- 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] How to dial multiple extensions at once like in aring group and put them in conference?
Zeeshan Zakaria escribió: > I think I should try the .call files. I haven't tried them in this > particular scenario yet. > > Miguel, what exactly you mean by calling asynchronously? I do > 'originate' in a 'while' loop once I have retrieved all the extensions > to dial from the database. I mean using the originate action with the Async option enabled, as explained here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate I wonder if you are waiting for a response (Synchronous) or if you are using that option (Asynchronous), which is clearly faster. However, the call file solution Danny proposed would work pretty well in your case. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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 interconnection problem
Lacking any response I tried to set "insecure=invite" on both sides. And lo and behold, the call gets through. Now, is this good or bad? /R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?
If you aren't doing an explicit async: true, then you are synchronous. Heed this post as well (Thanks Miguel) http://www.mail-archive.com/asterisk-users@lists.digium.com/msg231570.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Tuesday, October 27, 2009 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference? Zeeshan Zakaria escribió: > I think I should try the .call files. I haven't tried them in this > particular scenario yet. > > Miguel, what exactly you mean by calling asynchronously? I do > 'originate' in a 'while' loop once I have retrieved all the extensions > to dial from the database. I mean using the originate action with the Async option enabled, as explained here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate I wonder if you are waiting for a response (Synchronous) or if you are using that option (Asynchronous), which is clearly faster. However, the call file solution Danny proposed would work pretty well in your case. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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] SIP interconnection problem
Since you are doing peer-to-peer, this should be harmless. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Bielik Sent: Tuesday, October 27, 2009 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP interconnection problem Lacking any response I tried to set "insecure=invite" on both sides. And lo and behold, the call gets through. Now, is this good or bad? /R ___ -- 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 as the recording server for Avaya Definity
> Research wrote: >> . I saw a nice article on voip-info.org on how to replace voicemail >> server for Avaya Definity with asterisk. >> > > Could you send me the link of the article? I'll be looking into doing > this within the next year. > > Thanks, > > Doug Hi Doug See: http://www.voip-info.org/wiki/view/Asterisk-Partner+ACS+for+Voicemail The problem i have is how to use this info to replace nice/witness recording server Sam ___ -- 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] Codecs with MixMonitor (,a) option
Hi all, Another simple question: does it make sense to use the append option in MixMonitor (,a) when the codec is gsm? Or it works only when the codec is an uncompressed one like ulaw, alaw or slin? Thanks, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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] The Mobile devices are not able to register on my asterisk
Dear All; I am facing a problem that all the mobile devices that support SIP and are able to register with a lot of providers, they are not able to register on my asterisk. What could be the reason? Any specific thing I have to do? The used port is UDP 5060 Actually, any SIP Phone can register with my asterisk, but when I try from the mobile devices, it does not !! (While these mobiles are able to register with other SIP service provider). Any help? Regards Bilal ___ -- 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 Mobile devices are not able to register on myasterisk
Try this link http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Netwo rk -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 27, 2009 3:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] The Mobile devices are not able to register on myasterisk Dear All; I am facing a problem that all the mobile devices that support SIP and are able to register with a lot of providers, they are not able to register on my asterisk. What could be the reason? Any specific thing I have to do? The used port is UDP 5060 Actually, any SIP Phone can register with my asterisk, but when I try from the mobile devices, it does not !! (While these mobiles are able to register with other SIP service provider). Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Software for PC-PC voice comunication
I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help ___ -- 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] Software for PC-PC voice comunication
Xlite softphone?? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of giancarlo lombardo Sent: Tuesday, October 27, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Software for PC-PC voice comunication I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help ___ -- 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] Installing Asterisk
On Tue, 2009-10-27 at 09:34 +, aster...@opensourcesolution.in wrote: > hi , > > i have started reading asterisk book need your guidance.friend as i > am newbie in asterisk so plz plz forgive me if i ask stupid questions. > > > > Installing Asterisk > > - on which linux flavour i should start the installation of asterisk > (CentOs,Fedora,Ubuntu) > right now i am using ubuntu 8.04lts > > - What are the essential packages. > > - whats is PRI, BRI interfaces > > - what is asterisk now > > > > Regards, > > Pawan Which book? ___ -- 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] Software for PC-PC voice comunication
On Tue, 27 Oct 2009, giancarlo lombardo wrote: > I just installed an Asterisknow server can someone suggest a software to > be used for a PC - PC voice comunication to test in easy way the > functionalities of my server. If your PC is running Windows, DIAX is the smallest and easiest soft phone -- no installation, uses IAX instead of SIP. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] need to find firmware for cisco ata-188
Hi there, I have an old Cisco ATA-188-I2-A that I want to revive but with SCCP (right now it has SIP). the version i am looking for is ata_03_02_04_sccp_090202_a.zip i want to do a home experiment with chan_sccp and some recompilations any links beside cisco to download the firmware? i do not have a valid contract, so cisco does not allow me to download it. thanks in advance. erick. ___ -- 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] need to find firmware for cisco ata-188
On 27 Oct 2009, at 23:29, Erick Perez wrote: > any links beside cisco to download the firmware? > i do not have a valid contract, so cisco does not allow me to > download it. So you want to pirate it instead? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?
"Async: True" was the solutions to my problem. Thanks for pointing me in this direction. -- Zeeshan A Zakaria On Tue, Oct 27, 2009 at 12:11 PM, Danny Nicholas wrote: > If you aren't doing an explicit async: true, then you are synchronous. > Heed > this post as well (Thanks Miguel) > http://www.mail-archive.com/asterisk-users@lists.digium.com/msg231570.html > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel > Molina > Sent: Tuesday, October 27, 2009 10:59 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] How to dial multiple extensions at once like > in aring group and put them in conference? > > Zeeshan Zakaria escribió: > > I think I should try the .call files. I haven't tried them in this > > particular scenario yet. > > > > Miguel, what exactly you mean by calling asynchronously? I do > > 'originate' in a 'while' loop once I have retrieved all the extensions > > to dial from the database. > I mean using the originate action with the Async option enabled, as > explained here: > > http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate > > I wonder if you are waiting for a response (Synchronous) or if you are > using that option (Asynchronous), which is clearly faster. However, the > call file solution Danny proposed would work pretty well in your case. > > Cheers, > > -- > Ing. Miguel Molina > Grupo de Tecnología > Millenium Phone Center > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confusion on caller-ID with SIP provider
If I have a SIP "provider" (in this case a PBX using SIP trunks), and I want to send the calling extension number and name as the "from" in the SIP invite, how do I set up my sip.conf entry for that provider? I find the documentation confusing on this point. ___ -- 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] Confusion on caller-ID with SIP provider
callerid=Some Name In From Header <7065551212> Richard Kenner wrote: > If I have a SIP "provider" (in this case a PBX using SIP trunks), and > I want to send the calling extension number and name as the "from" in > the SIP invite, how do I set up my sip.conf entry for that provider? I > find the documentation confusing on this point. > > ___ > -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] Confusion on caller-ID with SIP provider
> callerid=Some Name In From Header <7065551212> So the first part is the NAME and the second the number, right? But my question was how to have that be information from the CALLERID channel variable rather than a fixed value in sip.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once likein aring group and put them in conference?
On 28/10/09 3:52 AM, Danny Nicholas wrote: > This might be a better application of a call file than an AMI originate. > The AMI originate in this case has to operate in a threaded fashion, > whereas if you created a call file for each extension and dumped them > into /var/spool/asterisk/outgoing, pbx.c would call all of them at once > without the “first pickup” problem. Not true - you can use Async mode in an Asterisk Manager originate command to create a call and return instantly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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/Cisco AS5300 => Two problems in incoming (extension not found)
Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. (042600 = my phone number) First problems: Why he don't see the extension ? sip.conf: [AS5300] host=192.168.50.125 context=as5300-incoming type=peer dtmf=rfc2833 nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp [as5300-incoming] exten => 042600,1,Ringing exten => 042600,2,Answer exten => 042600,3,Dial(SIP/Jpc,25,m) exten => 042600,4,Hangup And second problems: "Call from '' to", AS5300 don't sent the number of the caller ? Thanks for your help Jerome ___ -- 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] Dialing out a T1
Ok, so this might seem like a stupid question, but I don't quite understand how to dial out to the pstn though my T1 from a specific number. Maybe i'm missing something, but everything I'm reading has you dial a number from the "group" but that's not what i'm looking for. If someone can just point me into the right direction, I would greatly appreciate it. Thanks ~T ___ -- 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/Cisco AS5300 => Two problems in incoming (extension not found)
Phibee Network Operation Center a écrit : > Hi > > Now, my Cisco AS5300 sent call to my asterisk, but two problems: > > When i call the phone number, i have: > > [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: > Call from '' to extension '042600' rejected because extension not found. > [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: > Call from '' to extension '042600' rejected because extension not found. > > (042600 = my phone number) > <..> > I have put a debug: [Kvoip*CLI> <--- SIP read from UDP://192.168.50.125:59124 ---> INVITE sip:0426000...@192.168.50.130:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.125:5060 From: ;tag=6950F0-25C7 To: Date: Wed, 28 Oct 2009 05:16:26 GMT Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3761097657-3266777566-2192416711-2957366127 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1256706986 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 8642 2741 IN IP4 192.168.50.125 s=SIP Call c=IN IP4 192.168.50.125 t=0 0 m=audio 18726 RTP/AVP 8 101 c=IN IP4 192.168.50.125 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-> [Kvoip*CLI> --- (20 headers 11 lines) --- [Kvoip*CLI> Sending to 192.168.50.125 : 5060 (no NAT) [Kvoip*CLI> Using INVITE request as basis request - e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 [Kvoip*CLI> No matching peer for '47700' from '192.168.50.125:59124' [Kvoip*CLI> Found RTP audio format 8 [Kvoip*CLI> Found RTP audio format 101 [Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI> Found audio description format PCMA for ID 8 [Kvoip*CLI> Found audio description format telephone-event for ID 101 [Kvoip*CLI> Got unsupported a:fmtp in SDP offer [Kvoip*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Kvoip*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI> Looking for 042600 in default (domain 192.168.50.130) [Kvoip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.50.125:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.50.125:5060;received=192.168.50.125 From: ;tag=6950F0-25C7 To: ;tag=as25696e60 Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Ok, i see that: 1- Cisco sent the phone number of the caller (47700) 2- I have a "To: " 192.168.50.130 = My Asterisk Server 192.168.50.125 = My Cisco AS5300 3- i have a "No matching peer for '47700' from '192.168.50.125:59124'" why he search a peer with "47700" ?? bye Jerome ___ -- 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] Dialing out a T1
On Oct 27, 2009, at 10:50 PM, trebaum wrote: > Ok, so this might seem like a stupid question, but I don't quite > understand how to dial out to the pstn though my T1 from a specific > number. Maybe i'm missing something, but everything I'm reading has > you dial a number from the "group" but that's not what i'm looking > for. If someone can just point me into the right direction, I would > greatly appreciate it. > Thanks > > ~T Ok, so I was able to get my answer from the IRC channel. Just in case anyone is curious in the future, you just need to set the CID to the number you want to call from as you are making the outgoing call. ~T ___ -- 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 client MAC address.
hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- 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 client MAC address.
>From Linux you could use arp | grep "192.168.0.1" substituting the IP address of the SIP device. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- 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 client MAC address.
hello david, what in case of sip client is behind "NAT", and i want SIP client IP address. not from system from which client registered. if it is a SIP phone then what? if you have any idea then tell me. regards dhaval On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C < david.klavers...@intergraph.com> wrote: > From Linux you could use > > > > arp | grep "192.168.0.1" > > > > substituting the IP address of the SIP device. > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA > *Sent:* Wednesday, 28 October 2009 4:29 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] SIP client MAC address. > > > > hello, > > is there any facility to get SIP client (ex. softphone,ipphone) MAC address > on asterisk. > > based on that we authenticated client in anyway. > > i tried with sip debug but i didn't got any MAC address related field in > all packets. > > > > regards > Dhaval > > ___ > -- 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] SIP client MAC address.
If there is more than one SIP devices operating from the same NAT device then I'm not sure what you could do as it would always show the same IP for all SIP devices behind the same NAT. If there is only one device behind that NAT making a connection to your server then that is easy, if not I think your screwed. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP client MAC address. hello david, what in case of sip client is behind "NAT", and i want SIP client IP address. not from system from which client registered. if it is a SIP phone then what? if you have any idea then tell me. regards dhaval On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C wrote: >From Linux you could use arp | grep "192.168.0.1" substituting the IP address of the SIP device. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- 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
